Network Working Group J. Uberti
Internet-Draft Google
Intended status: Standards Track C. Jennings
Expires: April 24, 2017 Cisco
E. Rescorla, Ed.
Mozilla
October 21, 2016
Javascript Session Establishment Protocoldraft-ietf-rtcweb-jsep-17
Abstract
This document describes the mechanisms for allowing a Javascript
application to control the signaling plane of a multimedia session
via the interface specified in the W3C RTCPeerConnection API, and
discusses how this relates to existing signaling protocols.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on April 24, 2017.
Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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publication of this document. Please review these documents
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include Simplified BSD License text as described in Section 4.e of
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Internet-Draft JSEP October 20161. Introduction
This document describes how the W3C WEBRTC RTCPeerConnection
interface [W3C.WD-webrtc-20140617] is used to control the setup,
management and teardown of a multimedia session.
1.1. General Design of JSEP
The thinking behind WebRTC call setup has been to fully specify and
control the media plane, but to leave the signaling plane up to the
application as much as possible. The rationale is that different
applications may prefer to use different protocols, such as the
existing SIP or Jingle call signaling protocols, or something custom
to the particular application, perhaps for a novel use case. In this
approach, the key information that needs to be exchanged is the
multimedia session description, which specifies the necessary
transport and media configuration information necessary to establish
the media plane.
With these considerations in mind, this document describes the
Javascript Session Establishment Protocol (JSEP) that allows for full
control of the signaling state machine from Javascript. JSEP removes
the browser almost entirely from the core signaling flow, which is
instead handled by the Javascript making use of two interfaces: (1)
passing in local and remote session descriptions and (2) interacting
with the ICE state machine.
In this document, the use of JSEP is described as if it always occurs
between two browsers. Note though in many cases it will actually be
between a browser and some kind of server, such as a gateway or MCU.
This distinction is invisible to the browser; it just follows the
instructions it is given via the API.
JSEP's handling of session descriptions is simple and
straightforward. Whenever an offer/answer exchange is needed, the
initiating side creates an offer by calling a createOffer() API. The
application optionally modifies that offer, and then uses it to set
up its local config via the setLocalDescription() API. The offer is
then sent off to the remote side over its preferred signaling
mechanism (e.g., WebSockets); upon receipt of that offer, the remote
party installs it using the setRemoteDescription() API.
To complete the offer/answer exchange, the remote party uses the
createAnswer() API to generate an appropriate answer, applies it
using the setLocalDescription() API, and sends the answer back to the
initiator over the signaling channel. When the initiator gets that
answer, it installs it using the setRemoteDescription() API, and
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initial setup is complete. This process can be repeated for
additional offer/answer exchanges.
Regarding ICE [RFC5245], JSEP decouples the ICE state machine from
the overall signaling state machine, as the ICE state machine must
remain in the browser, because only the browser has the necessary
knowledge of candidates and other transport info. Performing this
separation also provides additional flexibility; in protocols that
decouple session descriptions from transport, such as Jingle, the
session description can be sent immediately and the transport
information can be sent when available. In protocols that don't,
such as SIP, the information can be used in the aggregated form.
Sending transport information separately can allow for faster ICE and
DTLS startup, since ICE checks can start as soon as any transport
information is available rather than waiting for all of it.
Through its abstraction of signaling, the JSEP approach does require
the application to be aware of the signaling process. While the
application does not need to understand the contents of session
descriptions to set up a call, the application must call the right
APIs at the right times, convert the session descriptions and ICE
information into the defined messages of its chosen signaling
protocol, and perform the reverse conversion on the messages it
receives from the other side.
One way to mitigate this is to provide a Javascript library that
hides this complexity from the developer; said library would
implement a given signaling protocol along with its state machine and
serialization code, presenting a higher level call-oriented interface
to the application developer. For example, libraries exist to adapt
the JSEP API into an API suitable for a SIP or XMPP. Thus, JSEP
provides greater control for the experienced developer without
forcing any additional complexity on the novice developer.
1.2. Other Approaches Considered
One approach that was considered instead of JSEP was to include a
lightweight signaling protocol. Instead of providing session
descriptions to the API, the API would produce and consume messages
from this protocol. While providing a more high-level API, this put
more control of signaling within the browser, forcing the browser to
have to understand and handle concepts like signaling glare. In
addition, it prevented the application from driving the state machine
to a desired state, as is needed in the page reload case.
A second approach that was considered but not chosen was to decouple
the management of the media control objects from session
descriptions, instead offering APIs that would control each component
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directly. This was rejected based on a feeling that requiring
exposure of this level of complexity to the application programmer
would not be beneficial; it would result in an API where even a
simple example would require a significant amount of code to
orchestrate all the needed interactions, as well as creating a large
API surface that needed to be agreed upon and documented. In
addition, these API points could be called in any order, resulting in
a more complex set of interactions with the media subsystem than the
JSEP approach, which specifies how session descriptions are to be
evaluated and applied.
One variation on JSEP that was considered was to keep the basic
session description-oriented API, but to move the mechanism for
generating offers and answers out of the browser. Instead of
providing createOffer/createAnswer methods within the browser, this
approach would instead expose a getCapabilities API which would
provide the application with the information it needed in order to
generate its own session descriptions. This increases the amount of
work that the application needs to do; it needs to know how to
generate session descriptions from capabilities, and especially how
to generate the correct answer from an arbitrary offer and the
supported capabilities. While this could certainly be addressed by
using a library like the one mentioned above, it basically forces the
use of said library even for a simple example. Providing
createOffer/createAnswer avoids this problem, but still allows
applications to generate their own offers/answers (to a large extent)
if they choose, using the description generated by createOffer as an
indication of the browser's capabilities.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
3. Semantics and Syntax3.1. Signaling Model
JSEP does not specify a particular signaling model or state machine,
other than the generic need to exchange session descriptions in the
fashion described by [RFC3264](offer/answer) in order for both sides
of the session to know how to conduct the session. JSEP provides
mechanisms to create offers and answers, as well as to apply them to
a session. However, the browser is totally decoupled from the actual
mechanism by which these offers and answers are communicated to the
remote side, including addressing, retransmission, forking, and glare
handling. These issues are left entirely up to the application; the
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application has complete control over which offers and answers get
handed to the browser, and when.
+-----------+ +-----------+
| Web App |<--- App-Specific Signaling -->| Web App |
+-----------+ +-----------+
^ ^
| SDP | SDP
V V
+-----------+ +-----------+
| Browser |<----------- Media ------------>| Browser |
+-----------+ +-----------+
Figure 1: JSEP Signaling Model
3.2. Session Descriptions and State Machine
In order to establish the media plane, the user agent needs specific
parameters to indicate what to transmit to the remote side, as well
as how to handle the media that is received. These parameters are
determined by the exchange of session descriptions in offers and
answers, and there are certain details to this process that must be
handled in the JSEP APIs.
Whether a session description applies to the local side or the remote
side affects the meaning of that description. For example, the list
of codecs sent to a remote party indicates what the local side is
willing to receive, which, when intersected with the set of codecs
the remote side supports, specifies what the remote side should send.
However, not all parameters follow this rule; for example, the DTLS-
SRTP parameters [RFC5763] sent to a remote party indicate what
certificate the local side will use in DTLS setup, and thereby what
the remote party should expect to receive; the remote party will have
to accept these parameters, with no option to choose different
values.
In addition, various RFCs put different conditions on the format of
offers versus answers. For example, an offer may propose an
arbitrary number of media streams (i.e. m= sections), but an answer
must contain the exact same number as the offer.
Lastly, while the exact media parameters are only known only after an
offer and an answer have been exchanged, it is possible for the
offerer to receive media after they have sent an offer and before
they have received an answer. To properly process incoming media in
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this case, the offerer's media handler must be aware of the details
of the offer before the answer arrives.
Therefore, in order to handle session descriptions properly, the user
agent needs:
1. To know if a session description pertains to the local or remote
side.
2. To know if a session description is an offer or an answer.
3. To allow the offer to be specified independently of the answer.
JSEP addresses this by adding both setLocalDescription and
setRemoteDescription methods and having session description objects
contain a type field indicating the type of session description being
supplied. This satisfies the requirements listed above for both the
offerer, who first calls setLocalDescription(sdp [offer]) and then
later setRemoteDescription(sdp [answer]), as well as for the
answerer, who first calls setRemoteDescription(sdp [offer]) and then
later setLocalDescription(sdp [answer]).
JSEP also allows for an answer to be treated as provisional by the
application. Provisional answers provide a way for an answerer to
communicate initial session parameters back to the offerer, in order
to allow the session to begin, while allowing a final answer to be
specified later. This concept of a final answer is important to the
offer/answer model; when such an answer is received, any extra
resources allocated by the caller can be released, now that the exact
session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders.
Provisional answers, on the other hand, do no such deallocation
results; as a result, multiple dissimilar provisional answers can be
received and applied during call setup.
In [RFC3264], the constraint at the signaling level is that only one
offer can be outstanding for a given session, but at the media stack
level, a new offer can be generated at any point. For example, when
using SIP for signaling, if one offer is sent, then cancelled using a
SIP CANCEL, another offer can be generated even though no answer was
received for the first offer. To support this, the JSEP media layer
can provide an offer via the createOffer() method whenever the
Javascript application needs one for the signaling. The answerer can
send back zero or more provisional answers, and finally end the
offer-answer exchange by sending a final answer. The state machine
for this is as follows:
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Internet-Draft JSEP October 20163.3. Session Description Format
In the WebRTC specification, session descriptions are formatted as
SDP messages. While this format is not optimal for manipulation from
Javascript, it is widely accepted, and frequently updated with new
features. Any alternate encoding of session descriptions would have
to keep pace with the changes to SDP, at least until the time that
this new encoding eclipsed SDP in popularity. As a result, JSEP
currently uses SDP as the internal representation for its session
descriptions.
However, to simplify Javascript processing, and provide for future
flexibility, the SDP syntax is encapsulated within a
SessionDescription object, which can be constructed from SDP, and be
serialized out to SDP. If future specifications agree on a JSON
format for session descriptions, we could easily enable this object
to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript. In the
meantime, Javascript libraries can be used to perform these
manipulations.
Note that most applications should be able to treat the
SessionDescriptions produced and consumed by these various API calls
as opaque blobs; that is, the application will not need to read or
change them.
3.4. Session Description Control
In order to give the application control over various common session
parameters, JSEP provides control surfaces which tell the browser how
to generate session descriptions. This avoids the need for
Javascript to modify session descriptions in most cases.
Changes to these objects result in changes to the session
descriptions generated by subsequent createOffer/Answer calls.
3.4.1. RtpTransceivers
RtpTransceivers allow the application to control the RTP media
associated with one m= section. Each RtpTransceiver has an RtpSender
and an RtpReceiver, which an application can use to control the
sending and receiving of RTP media. The application may also modify
the RtpTransceiver directly, for instance, by stopping it.
RtpTransceivers generally have a 1:1 mapping with m= sections,
although there may be more RtpTransceivers than m= sections when
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RtpTransceivers are created but not yet associated with a m= section,
or if RtpTransceivers have been stopped and disassociated from m=
sections. An RtpTransceiver is never associated with more than one
m= section, and once a session description is applied, a m= section
is always associated with exactly one RtpTransceiver.
RtpTransceivers can be created explicitly by the application or
implicitly by calling setRemoteDescription with an offer that adds
new m= sections.
3.4.2. RtpSenders
RtpSenders allow the application to control how RTP media is sent.
3.4.3. RtpReceivers
RtpReceivers allows the application to control how RTP media is
received.
3.5. ICE3.5.1. ICE Gathering Overview
JSEP gathers ICE candidates as needed by the application. Collection
of ICE candidates is referred to as a gathering phase, and this is
triggered either by the addition of a new or recycled m= line to the
local session description, or new ICE credentials in the description,
indicating an ICE restart. Use of new ICE credentials can be
triggered explicitly by the application, or implicitly by the browser
in response to changes in the ICE configuration.
When the ICE configuration changes in a way that requires a new
gathering phase, a 'needs-ice-restart' bit is set. When this bit is
set, calls to the createOffer API will generate new ICE credentials.
This bit is cleared by a call to the setLocalDescription API with new
ICE credentials from either an offer or an answer, i.e., from either
a local- or remote-initiated ICE restart.
When a new gathering phase starts, the ICE Agent will notify the
application that gathering is occurring through an event. Then, when
each new ICE candidate becomes available, the ICE Agent will supply
it to the application via an additional event; these candidates will
also automatically be added to the current and/or pending local
session description. Finally, when all candidates have been
gathered, an event will be dispatched to signal that the gathering
process is complete.
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Note that gathering phases only gather the candidates needed by
new/recycled/restarting m= lines; other m= lines continue to use
their existing candidates. Also, when bundling is active, candidates
are only gathered (and exchanged) for the m= lines referenced in
BUNDLE-tags, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation].
3.5.2. ICE Candidate Trickling
Candidate trickling is a technique through which a caller may
incrementally provide candidates to the callee after the initial
offer has been dispatched; the semantics of "Trickle ICE" are defined
in [I-D.ietf-ice-trickle]. This process allows the callee to begin
acting upon the call and setting up the ICE (and perhaps DTLS)
connections immediately, without having to wait for the caller to
gather all possible candidates. This results in faster media setup
in cases where gathering is not performed prior to initiating the
call.
JSEP supports optional candidate trickling by providing APIs, as
described above, that provide control and feedback on the ICE
candidate gathering process. Applications that support candidate
trickling can send the initial offer immediately and send individual
candidates when they get the notified of a new candidate;
applications that do not support this feature can simply wait for the
indication that gathering is complete, and then create and send their
offer, with all the candidates, at this time.
Upon receipt of trickled candidates, the receiving application will
supply them to its ICE Agent. This triggers the ICE Agent to start
using the new remote candidates for connectivity checks.
3.5.2.1. ICE Candidate Format
As with session descriptions, the syntax of the IceCandidate object
provides some abstraction, but can be easily converted to and from
the SDP candidate lines.
The candidate lines are the only SDP information that is contained
within IceCandidate, as they represent the only information needed
that is not present in the initial offer (i.e., for trickle
candidates). This information is carried with the same syntax as the
"candidate-attribute" field defined for ICE. For example:
candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
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The IceCandidate object also contains fields to indicate which m=
line it should be associated with. The m= line can be identified in
one of two ways; either by a m= line index, or a MID. The m= line
index is a zero-based index, with index N referring to the N+1th m=
line in the SDP sent by the entity which sent the IceCandidate. The
MID uses the "media stream identification" attribute, as defined in
[RFC5888], Section 4, to identify the m= line. JSEP implementations
creating an ICE Candidate object MUST populate both of these fields,
using the MID of the associated RtpTransceiver object (which may be
locally generated by the answerer when interacting with a non-JSEP
remote endpoint that does not support the MID attribute, as discussed
in Section 5.9 below). Implementations receiving an ICE Candidate
object MUST use the MID if present, or the m= line index, if not (the
non-JSEP remote endpoint case).
3.5.3. ICE Candidate Policy
Typically, when gathering ICE candidates, the browser will gather all
possible forms of initial candidates - host, server reflexive, and
relay. However, in certain cases, applications may want to have more
specific control over the gathering process, due to privacy or
related concerns. For example, one may want to suppress the use of
host candidates, to avoid exposing information about the local
network, or go as far as only using relay candidates, to leak as
little location information as possible (note that these choices come
with corresponding operational costs). To accomplish this, the
browser MUST allow the application to restrict which ICE candidates
are used in a session. Note that this filtering is applied on top of
any restrictions the browser chooses to enforce regarding which IP
addresses are permitted for the application, as discussed in
[I-D.ietf-rtcweb-ip-handling].
There may also be cases where the application wants to change which
types of candidates are used while the session is active. A prime
example is where a callee may initially want to use only relay
candidates, to avoid leaking location information to an arbitrary
caller, but then change to use all candidates (for lower operational
cost) once the user has indicated they want to take the call. For
this scenario, the browser MUST allow the candidate policy to be
changed in mid-session, subject to the aforementioned interactions
with local policy.
To administer the ICE candidate policy, the browser will determine
the current setting at the start of each gathering phase. Then,
during the gathering phase, the browser MUST NOT expose candidates
disallowed by the current policy to the application, use them as the
source of connectivity checks, or indirectly expose them via other
fields, such as the raddr/rport attributes for other ICE candidates.
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Later, if a different policy is specified by the application, the
application can apply it by kicking off a new gathering phase via an
ICE restart.
3.5.4. ICE Candidate Pool
JSEP applications typically inform the browser to begin ICE gathering
via the information supplied to setLocalDescription, as this is where
the app specifies the number of media streams, and thereby ICE
components, for which to gather candidates. However, to accelerate
cases where the application knows the number of ICE components to use
ahead of time, it may ask the browser to gather a pool of potential
ICE candidates to help ensure rapid media setup.
When setLocalDescription is eventually called, and the browser goes
to gather the needed ICE candidates, it SHOULD start by checking if
any candidates are available in the pool. If there are candidates in
the pool, they SHOULD be handed to the application immediately via
the ICE candidate event. If the pool becomes depleted, either
because a larger-than-expected number of ICE components is used, or
because the pool has not had enough time to gather candidates, the
remaining candidates are gathered as usual.
One example of where this concept is useful is an application that
expects an incoming call at some point in the future, and wants to
minimize the time it takes to establish connectivity, to avoid
clipping of initial media. By pre-gathering candidates into the
pool, it can exchange and start sending connectivity checks from
these candidates almost immediately upon receipt of a call. Note
though that by holding on to these pre-gathered candidates, which
will be kept alive as long as they may be needed, the application
will consume resources on the STUN/TURN servers it is using.
3.6. Video Size Negotiation
Video size negotiation is the process through which a receiver can
use the "a=imageattr" SDP attribute [RFC6236] to indicate what video
frame sizes it is capable of receiving. A receiver may have hard
limits on what its video decoder can process, or it may wish to
constrain what it receives due to application preferences, e.g. a
specific size for the window in which the video will be displayed.
Note that certain codecs support transmission of samples with aspect
ratios other than 1.0 (i.e., non-square pixels). JSEP
implementations will not transmit non-square pixels, but SHOULD
receive and render such video with the correct aspect ratio.
However, sample aspect ratio has no impact on the size negotiation
described below; all dimensions assume square pixels.
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In order to determine the limits on what video resolution a receiver
wants to receive, it will intersect its decoder hard limits with any
mandatory constraints that have been applied to the associated
MediaStreamTrack. If the decoder limits are unknown, e.g. when using
a software decoder, the mandatory constraints are used directly. For
the answerer, these mandatory constraints can be applied to the
remote MediaStreamTracks that are created by a setRemoteDescription
call, and will affect the output of the ensuing createAnswer call.
Any constraints set after setLocalDescription is used to set the
answer will result in a new offer-answer exchange. For the offerer,
because it does not know about any remote MediaStreamTracks until it
receives the answer, the offer can only reflect decoder hard limits.
If the offerer wishes to set mandatory constraints on video
resolution, it must do so after receiving the answer, and the result
will be a new offer-answer to communicate them.
If there are no known decoder limits or mandatory constraints, the
"a=imageattr" attribute SHOULD be omitted.
Otherwise, an "a=imageattr" attribute is created with "recv"
direction, and the resulting resolution space formed by intersecting
the decoder limits and constraints is used to specify its minimum and
maximum x= and y= values. If the intersection is the null set, i.e.,
there are no resolutions that are permitted by both the decoder and
the mandatory constraints, this SHOULD be represented by x=0 and y=0
values.
The rules here express a single set of preferences, and therefore,
the "a=imageattr" q= value is not important. It SHOULD be set to
1.0.
The "a=imageattr" field is payload type specific. When all video
codecs supported have the same capabilities, use of a single
attribute, with the wildcard payload type (*), is RECOMMENDED.
However, when the supported video codecs have differing capabilities,
specific "a=imageattr" attributes MUST be inserted for each payload
type.
As an example, consider a system with a HD-capable, multiformat video
decoder, where the application has constrained the received track to
at most 360p. In this case, the implementation would generate this
attribute:
a=imageattr:* recv [x=[16:640],y=[16:360],q=1.0]
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This declaration indicates that the receiver is capable of decoding
any image resolution from 16x16 up to 640x360 pixels.
3.6.2. Interpreting an imageattr Attribute
[RFC6236] defines "a=imageattr" to be an advisory field. This means
that it does not absolutely constrain the video formats that the
sender can use, but gives an indication of the preferred values.
This specification prescribes more specific behavior. When a sender
of a given MediaStreamTrack, which is producing video of a certain
resolution, receives an "a=imageattr recv" attribute, it MUST check
to see if the original resolution meets the size criteria specified
in the attribute, and adapt the resolution accordingly by scaling (if
appropriate). Note that when considering a MediaStreamTrack that is
producing rotated video, the unrotated resolution MUST be used. This
is required regardless of whether the receiver supports performing
receive-side rotation (e.g., through CVO), as it significantly
simplifies the matching logic.
For the purposes of resolution negotiation, only size limits are
considered. Any other values, e.g. picture or sample aspect ratio,
MUST be ignored.
When communicating with a non-JSEP endpoint, multiple relevant
"a=imageattr recv" attributes may be received. If this occurs,
attributes other than the one with the highest "q=" value MUST be
ignored.
If an "a=imageattr recv" attribute references a different video codec
than what has been selected for the MediaStreamTrack, it MUST be
ignored.
If the original resolution matches the size limits in the attribute,
the track MUST be transmitted untouched.
If the original resolution exceeds the size limits in the attribute,
the sender SHOULD apply downscaling to the output of the
MediaStreamTrack in order to satisfy the limits. Downscaling MUST
NOT change the track aspect ratio.
If the original resolution is less than the size limits in the
attribute, upscaling is needed, but this may not be appropriate in
all cases. To address this concern, the application can set an
upscaling policy for each sent track. For this case, if upscaling is
permitted by policy, the sender SHOULD apply upscaling in order to
provide the desired resolution. Otherwise, the sender MUST NOT apply
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upscaling. The sender SHOULD NOT upscale in other cases, even if the
policy permits it. Upscaling MUST NOT change the track aspect ratio.
If there is no appropriate and permitted scaling mechanism that
allows the received size limits to be satisfied, the sender MUST NOT
transmit the track.
If the attribute includes a "sar=" (sample aspect ratio) value set to
something other than "1.0", indicating the receiver wants to receive
non-square pixels, this cannot be satisfied and the sender MUST NOT
transmit the track.
In the special case of receiving a maximum resolution of [0, 0], as
described above, the sender MUST NOT transmit the track.
3.7. Simulcast
JSEP supports simulcast of a MediaStreamTrack, where multiple
encodings of the source media can be transmitted within the context
of a single m= section. The current JSEP API is designed to allow
applications to send simulcasted media but only to receive a single
encoding. This allows for multi-user scenarios where each sending
client sends multiple encodings to a server, which then, for each
receiving client, chooses the appropriate encoding to forward.
Applications request support for simulcast by configuring multiple
encodings on an RTPSender, which, upon generation of an offer or
answer, are indicated in SDP markings on the corresponding m=
section, as described below. Receivers that understand simulcast and
are willing to receive it will also include SDP markings to indicate
their support, and JSEP endpoints will use these markings to
determine whether simulcast is permitted for a given RTPSender. If
simulcast support is not negotiated, the RTPSender will only use the
first configured encoding.
Note that the exact simulcast parameters are up to the sending
application. While the aforementioned SDP markings are provided to
ensure the remote side can receive and demux multiple simulcast
encodings, the specific resolutions and bitrates to be used for each
encoding are purely a send-side decision in JSEP.
JSEP currently does not provide an API to configure receipt of
simulcast. This means that if simulcast is offered by the remote
endpoint, the answer generated by a JSEP endpoint will not indicate
support for receipt of simulcast, and as such the remote endpoint
will only send a single encoding per m= section. In addition, when
the JSEP endpoint is the answerer, the permitted encodings for the
RTPSender must be consistent with the offer, but this information is
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currently not surfaced through any API. This means that established
simulcast streams will continue to work through a received re-offer,
but setting up initial simulcast by way of a received offer requires
out-of-band signaling or SDP inspection. Future versions of this
specification may add additional APIs to provide this control.
When using JSEP to transmit multiple encodings from a RTPSender, the
techniques from [I-D.ietf-mmusic-sdp-simulcast] and
[I-D.ietf-mmusic-rid] are used. Specifically, when multiple
encodings have been configured for a RTPSender, the m= section for
the RTPSender will include an "a=simulcast" attribute, as defined in
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2, with a "send" simulcast
stream description that lists each desired encoding, and no "recv"
simulcast stream description. The m= section will also include an
"a=rid" attribute for each encoding, as specfied in
[I-D.ietf-mmusic-rid], Section 4; the use of RID identifiers allows
the individual encodings to be disambiguated even though they are all
part of the same m= section.
3.8. Interactions With Forking
Some call signaling systems allow various types of forking where an
SDP Offer may be provided to more than one device. For example, SIP
[RFC3261] defines both a "Parallel Search" and "Sequential Search".
Although these are primarily signaling level issues that are outside
the scope of JSEP, they do have some impact on the configuration of
the media plane that is relevant. When forking happens at the
signaling layer, the Javascript application responsible for the
signaling needs to make the decisions about what media should be sent
or received at any point of time, as well as which remote endpoint it
should communicate with; JSEP is used to make sure the media engine
can make the RTP and media perform as required by the application.
The basic operations that the applications can have the media engine
do are:
o Start exchanging media with a given remote peer, but keep all the
resources reserved in the offer.
o Start exchanging media with a given remote peer, and free any
resources in the offer that are not being used.
3.8.1. Sequential Forking
Sequential forking involves a call being dispatched to multiple
remote callees, where each callee can accept the call, but only one
active session ever exists at a time; no mixing of received media is
performed.
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JSEP handles sequential forking well, allowing the application to
easily control the policy for selecting the desired remote endpoint.
When an answer arrives from one of the callees, the application can
choose to apply it either as a provisional answer, leaving open the
possibility of using a different answer in the future, or apply it as
a final answer, ending the setup flow.
In a "first-one-wins" situation, the first answer will be applied as
a final answer, and the application will reject any subsequent
answers. In SIP parlance, this would be ACK + BYE.
In a "last-one-wins" situation, all answers would be applied as
provisional answers, and any previous call leg will be terminated.
At some point, the application will end the setup process, perhaps
with a timer; at this point, the application could reapply the
pending remote description as a final answer.
3.8.2. Parallel Forking
Parallel forking involves a call being dispatched to multiple remote
callees, where each callee can accept the call, and multiple
simultaneous active signaling sessions can be established as a
result. If multiple callees send media at the same time, the
possibilities for handling this are described in Section 3.1 of
[RFC3960]. Most SIP devices today only support exchanging media with
a single device at a time, and do not try to mix multiple early media
audio sources, as that could result in a confusing situation. For
example, consider having a European ringback tone mixed together with
the North American ringback tone - the resulting sound would not be
like either tone, and would confuse the user. If the signaling
application wishes to only exchange media with one of the remote
endpoints at a time, then from a media engine point of view, this is
exactly like the sequential forking case.
In the parallel forking case where the Javascript application wishes
to simultaneously exchange media with multiple peers, the flow is
slightly more complex, but the Javascript application can follow the
strategy that [RFC3960] describes using UPDATE. The UPDATE approach
allows the signaling to set up a separate media flow for each peer
that it wishes to exchange media with. In JSEP, this offer used in
the UPDATE would be formed by simply creating a new PeerConnection
and making sure that the same local media streams have been added
into this new PeerConnection. Then the new PeerConnection object
would produce a SDP offer that could be used by the signaling to
perform the UPDATE strategy discussed in [RFC3960].
As a result of sharing the media streams, the application will end up
with N parallel PeerConnection sessions, each with a local and remote
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description and their own local and remote addresses. The media flow
from these sessions can be managed by specifying SDP direction
attributes in the descriptions, or the application can choose to play
out the media from all sessions mixed together. Of course, if the
application wants to only keep a single session, it can simply
terminate the sessions that it no longer needs.
4. Interface
This section details the basic operations that must be present to
implement JSEP functionality. The actual API exposed in the W3C API
may have somewhat different syntax, but should map easily to these
concepts.
4.1. PeerConnection4.1.1. Constructor
The PeerConnection constructor allows the application to specify
global parameters for the media session, such as the STUN/TURN
servers and credentials to use when gathering candidates, as well as
the initial ICE candidate policy and pool size, and also the bundle
policy to use.
If an ICE candidate policy is specified, it functions as described in
Section 3.5.3, causing the browser to only surface the permitted
candidates (including any internal browser filtering) to the
application, and only use those candidates for connectivity checks.
The set of available policies is as follows:
all: All candidates permitted by browser policy will be gathered and
used.
relay: All candidates except relay candidates will be filtered out.
This obfuscates the location information that might be ascertained
by the remote peer from the received candidates. Depending on how
the application deploys its relay servers, this could obfuscate
location to a metro or possibly even global level.
The default ICE candidate policy MUST be set to "all" as this is
generally the desired policy, and also typically reduces use of
application TURN server resources significantly.
If a size is specified for the ICE candidate pool, this indicates the
number of ICE components to pre-gather candidates for. Because pre-
gathering results in utilizing STUN/TURN server resources for
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potentially long periods of time, this must only occur upon
application request, and therefore the default candidate pool size
MUST be zero.
The application can specify its preferred policy regarding use of
bundle, the multiplexing mechanism defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation]. Regardless of policy, the
application will always try to negotiate bundle onto a single
transport, and will offer a single bundle group across all media
section; use of this single transport is contingent upon the answerer
accepting bundle. However, by specifying a policy from the list
below, the application can control exactly how aggressively it will
try to bundle media streams together, which affects how it will
interoperate with a non-bundle-aware endpoint. When negotiating with
a non-bundle-aware endpoint, only the streams not marked as bundle-
only streams will be established.
The set of available policies is as follows:
balanced: The first media section of each type (audio, video, or
application) will contain transport parameters, which will allow
an answerer to unbundle that section. The second and any
subsequent media section of each type will be marked bundle-only.
The result is that if there are N distinct media types, then
candidates will be gathered for for N media streams. This policy
balances desire to multiplex with the need to ensure basic audio
and video can still be negotiated in legacy cases. When acting as
answerer, if there is no bundle group in the offer, the
implementation will reject all but the first m= section of each
type.
max-compat: All media sections will contain transport parameters;
none will be marked as bundle-only. This policy will allow all
streams to be received by non-bundle-aware endpoints, but require
separate candidates to be gathered for each media stream.
max-bundle: Only the first media section will contain transport
parameters; all streams other than the first will be marked as
bundle-only. This policy aims to minimize candidate gathering and
maximize multiplexing, at the cost of less compatibility with
legacy endpoints. When acting as answerer, the implementation
will reject any m= sections other than the first m= section,
unless they are in the same bundle group as that m= section.
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As it provides the best tradeoff between performance and
compatibility with legacy endpoints, the default bundle policy MUST
be set to "balanced".
The application can specify its preferred policy regarding use of
RTP/RTCP multiplexing [RFC5761] using one of the following policies:
negotiate: The browser will gather both RTP and RTCP candidates but
also will offer "a=rtcp-mux", thus allowing for compatibility with
either multiplexing or non-multiplexing endpoints.
require: The browser will only gather RTP candidates. This halves
the number of candidates that the offerer needs to gather. When
acting as answerer, the implementation will reject any m= section
that does not contain an "a=rtcp-mux" attribute.
The default multiplexing policy MUST be set to "require".
Implementations MAY choose to reject attempts by the application to
set the multiplexing policy to "negotiate".
4.1.2. addTrack
The addTrack method adds a MediaStreamTrack to the PeerConnection,
using the MediaStream argument to associate the track with other
tracks in the same MediaStream, so that they can be added to the same
"LS" group when creating an offer or answer. addTrack attempts to
minimize the number of transceivers as follows: If the PeerConnection
is in the "have-remote-offer" state, the track will be attached to
the first compatible transceiver that was created by the most recent
call to setRemoteDescription() and does not have a local track.
Otherwise, a new transceiver will be created, as described in
Section 4.1.3.
4.1.3. addTransceiver
The addTransceiver method adds a new RTPTransceiver to the
PeerConnection. If a MediaStreamTrack argument is provided, then the
transceiver will be configured with that media type and the track
will be attached to the transceiver. Otherwise, the application MUST
explicitly specify the type; this mode is useful for creating
recvonly transceivers as well as for creating transceivers to which a
track can be attached at some later point.
At the time of creation, the application can also specify a
transceiver direction attribute, a set of MediaStreams which the
transceiver is associated with (allowing LS group assignments), and a
set of encodings for the media (used for simulcast as described in
Section 3.7).
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Internet-Draft JSEP October 20164.1.4. createDataChannel
The createDataChannel method creates a new data channel and attaches
it to the PeerConnection. If no data channel currently exists for
this PeerConnection, then a new offer/answer exchange is required.
All data channels on a given PeerConnection share the same SCTP/DTLS
association and therefore the same m= section, so subsequent creation
of data channels does not have any impact on the JSEP state.
The createDataChannel method also includes a number of arguments
which are used by the PeerConnection (e.g., maxPacketLifetime) but
are not reflected in the SDP and do not affect the JSEP state.
4.1.5. createOffer
The createOffer method generates a blob of SDP that contains a
[RFC3264] offer with the supported configurations for the session,
including descriptions of the media added to this PeerConnection, the
codec/RTP/RTCP options supported by this implementation, and any
candidates that have been gathered by the ICE Agent. An options
parameter may be supplied to provide additional control over the
generated offer. This options parameter allows an application to
trigger an ICE restart, for the purpose of reestablishing
connectivity.
In the initial offer, the generated SDP will contain all desired
functionality for the session (functionality that is supported but
not desired by default may be omitted); for each SDP line, the
generation of the SDP will follow the process defined for generating
an initial offer from the document that specifies the given SDP line.
The exact handling of initial offer generation is detailed in
Section 5.2.1 below.
In the event createOffer is called after the session is established,
createOffer will generate an offer to modify the current session
based on any changes that have been made to the session, e.g., adding
or stopping RtpTransceivers, or requesting an ICE restart. For each
existing stream, the generation of each SDP line must follow the
process defined for generating an updated offer from the RFC that
specifies the given SDP line. For each new stream, the generation of
the SDP must follow the process of generating an initial offer, as
mentioned above. If no changes have been made, or for SDP lines that
are unaffected by the requested changes, the offer will only contain
the parameters negotiated by the last offer-answer exchange. The
exact handling of subsequent offer generation is detailed in
Section 5.2.2. below.
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Session descriptions generated by createOffer must be immediately
usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an
offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those
resources. Because this method may need to inspect the system state
to determine the currently available resources, it may be implemented
as an async operation.
Calling this method may do things such as generate new ICE
credentials, but does not result in candidate gathering, or cause
media to start or stop flowing.
4.1.6. createAnswer
The createAnswer method generates a blob of SDP that contains a
[RFC3264] SDP answer with the supported configuration for the session
that is compatible with the parameters supplied in the most recent
call to setRemoteDescription, which MUST have been called prior to
calling createAnswer. Like createOffer, the returned blob contains
descriptions of the media added to this PeerConnection, the
codec/RTP/RTCP options negotiated for this session, and any
candidates that have been gathered by the ICE Agent. An options
parameter may be supplied to provide additional control over the
generated answer.
As an answer, the generated SDP will contain a specific configuration
that specifies how the media plane should be established; for each
SDP line, the generation of the SDP must follow the process defined
for generating an answer from the document that specifies the given
SDP line. The exact handling of answer generation is detailed in
Section 5.3. below.
Session descriptions generated by createAnswer must be immediately
usable by setLocalDescription; like createOffer, the returned
description should reflect the current state of the system. Because
this method may need to inspect the system state to determine the
currently available resources, it may need to be implemented as an
async operation.
Calling this method may do things such as generate new ICE
credentials, but does not trigger candidate gathering or change media
state.
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Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", "answer" or "rollback". These types provide
information as to how the description parameter should be parsed, and
how the media state should be changed.
"offer" indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A
description used as an "offer" may be applied anytime the
PeerConnection is in a stable state, or as an update to a previously
supplied but unanswered "offer".
"pranswer" indicates that a description should be parsed as an
answer, but not a final answer, and so should not result in the
freeing of allocated resources. It may result in the start of media
transmission, if the answer does not specify an inactive media
direction. A description used as a "pranswer" may be applied as a
response to an "offer", or an update to a previously sent "pranswer".
"answer" indicates that a description should be parsed as an answer,
the offer-answer exchange should be considered complete, and any
resources (decoders, candidates) that are no longer needed can be
released. A description used as an "answer" may be applied as a
response to an "offer", or an update to a previously sent "pranswer".
The only difference between a provisional and final answer is that
the final answer results in the freeing of any unused resources that
were allocated as a result of the offer. As such, the application
can use some discretion on whether an answer should be applied as
provisional or final, and can change the type of the session
description as needed. For example, in a serial forking scenario, an
application may receive multiple "final" answers, one from each
remote endpoint. The application could choose to accept the initial
answers as provisional answers, and only apply an answer as final
when it receives one that meets its criteria (e.g. a live user
instead of voicemail).
"rollback" is a special session description type implying that the
state machine should be rolled back to the previous stable state, as
described in Section 4.1.7.2. The contents MUST be empty.
4.1.7.1. Use of Provisional Answers
Most web applications will not need to create answers using the
"pranswer" type. While it is good practice to send an immediate
response to an "offer", in order to warm up the session transport and
prevent media clipping, the preferred handling for a web application
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would be to create and send an "inactive" final answer immediately
after receiving the offer. Later, when the called user actually
accepts the call, the application can create a new "sendrecv" offer
to update the previous offer/answer pair and start the media flow.
While this could also be done with an inactive "pranswer", followed
by a sendrecv "answer", the initial "pranswer" leaves the offer-
answer exchange open, which means that neither side can send an
updated offer during this time.
As an example, consider a typical web application that will set up a
data channel, an audio channel, and a video channel. When an
endpoint receives an offer with these channels, it could send an
answer accepting the data channel for two-way data, and accepting the
audio and video tracks as inactive or receive-only. It could then
ask the user to accept the call, acquire the local media streams, and
send a new offer to the remote side moving the audio and video to be
two-way media. By the time the human has accepted the call and
triggered the new offer, it is likely that the ICE and DTLS
handshaking for all the channels will already have finished.
Of course, some applications may not be able to perform this double
offer-answer exchange, particularly ones that are attempting to
gateway to legacy signaling protocols. In these cases, "pranswer"
can still provide the application with a mechanism to warm up the
transport.
4.1.7.2. Rollback
In certain situations it may be desirable to "undo" a change made to
setLocalDescription or setRemoteDescription. Consider a case where a
call is ongoing, and one side wants to change some of the session
parameters; that side generates an updated offer and then calls
setLocalDescription. However, the remote side, either before or
after setRemoteDescription, decides it does not want to accept the
new parameters, and sends a reject message back to the offerer. Now,
the offerer, and possibly the answerer as well, need to return to a
stable state and the previous local/remote description. To support
this, we introduce the concept of "rollback".
A rollback discards any proposed changes to the session, returning
the state machine to the stable state, and setting the pending local
and/or remote description back to null. Any resources or candidates
that were allocated by the abandoned local description are discarded;
any media that is received will be processed according to the
previous local and remote descriptions. Rollback can only be used to
cancel proposed changes; there is no support for rolling back from a
stable state to a previous stable state. Note that this implies that
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once the answerer has performed setLocalDescription with his answer,
this cannot be rolled back.
A rollback will disassociate any RtpTransceivers that were associated
with m= sections by the application of the rolled-back session
description (see Section 5.9 and Section 5.8). This means that some
RtpTransceivers that were previously associated will no longer be
associated with any m= section; in such cases, the value of the
RtpTransceiver's mid attribute MUST be set to null. RtpTransceivers
that were created by applying a remote offer that was subsequently
rolled back MUST be removed. However, a RtpTransceiver MUST NOT be
removed if the RtpTransceiver's RtpSender was activated by the
addTrack method. This is so that an application may call addTrack,
then call setRemoteDescription with an offer, then roll back that
offer, then call createOffer and have a m= section for the added
track appear in the generated offer.
A rollback is performed by supplying a session description of type
"rollback" with empty contents to either setLocalDescription or
setRemoteDescription, depending on which was most recently used (i.e.
if the new offer was supplied to setLocalDescription, the rollback
should be done using setLocalDescription as well).
4.1.8. setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply
the supplied session description as its local configuration. The
type field indicates whether the description should be processed as
an offer, provisional answer, or final answer; offers and answers are
checked differently, using the various rules that exist for each SDP
line.
This API changes the local media state; among other things, it sets
up local resources for receiving and decoding media. In order to
successfully handle scenarios where the application wants to offer to
change from one media format to a different, incompatible format, the
PeerConnection must be able to simultaneously support use of both the
current and pending local descriptions (e.g. support codecs that
exist in both descriptions) until a final answer is received, at
which point the PeerConnection can fully adopt the pending local
description, or roll back to the current description if the remote
side denied the change.
This API indirectly controls the candidate gathering process. When a
local description is supplied, and the number of transports currently
in use does not match the number of transports needed by the local
description, the PeerConnection will create transports as needed and
begin gathering candidates for them.
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If setRemoteDescription was previously called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media are available to
send, this will result in the starting of media transmission.
4.1.9. setRemoteDescription
The setRemoteDescription method instructs the PeerConnection to apply
the supplied session description as the desired remote configuration.
As in setLocalDescription, the type field of the description
indicates how it should be processed.
This API changes the local media state; among other things, it sets
up local resources for sending and encoding media.
If setLocalDescription was previously called with an offer, and
setRemoteDescription is called with an answer (provisional or final),
and the media directions are compatible, and media are available to
send, this will result in the starting of media transmission.
4.1.10. currentLocalDescription
The currentLocalDescription method returns a copy of the current
negotiated local description - i.e., the local description from the
last successful offer/answer exchange - in addition to any local
candidates that have been generated by the ICE Agent since the local
description was set.
A null object will be returned if an offer/answer exchange has not
yet been completed.
4.1.11. pendingLocalDescription
The pendingLocalDescription method returns a copy of the local
description currently in negotiation - i.e., a local offer set
without any corresponding remote answer - in addition to any local
candidates that have been generated by the ICE Agent since the local
description was set.
A null object will be returned if the state of the PeerConnection is
"stable" or "have-remote-offer".
4.1.12. currentRemoteDescription
The currentRemoteDescription method returns a copy of the current
negotiated remote description - i.e., the remote description from the
last successful offer/answer exchange - in addition to any remote
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candidates that have been supplied via processIceMessage since the
remote description was set.
A null object will be returned if an offer/answer exchange has not
yet been completed.
4.1.13. pendingRemoteDescription
The pendingRemoteDescription method returns a copy of the remote
description currently in negotiation - i.e., a remote offer set
without any corresponding local answer - in addition to any remote
candidates that have been supplied via processIceMessage since the
remote description was set.
A null object will be returned if the state of the PeerConnection is
"stable" or "have-local-offer".
4.1.14. canTrickleIceCandidates
The canTrickleIceCandidates property indicates whether the remote
side supports receiving trickled candidates. There are three
potential values:
null: No SDP has been received from the other side, so it is not
known if it can handle trickle. This is the initial value before
setRemoteDescription() is called.
true: SDP has been received from the other side indicating that it
can support trickle.
false: SDP has been received from the other side indicating that it
cannot support trickle.
As described in Section 3.5.2, JSEP implementations always provide
candidates to the application individually, consistent with what is
needed for Trickle ICE. However, applications can use the
canTrickleIceCandidates property to determine whether their peer can
actually do Trickle ICE, i.e., whether it is safe to send an initial
offer or answer followed later by candidates as they are gathered.
As "true" is the only value that definitively indicates remote
Trickle ICE support, an application which compares
canTrickleIceCandidates against "true" will by default attempt Half
Trickle on initial offers and Full Trickle on subsequent interactions
with a Trickle ICE-compatible agent.
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Internet-Draft JSEP October 20164.1.15. setConfiguration
The setConfiguration method allows the global configuration of the
PeerConnection, which was initially set by constructor parameters, to
be changed during the session. The effects of this method call
depend on when it is invoked, and differ depending on which specific
parameters are changed:
o Any changes to the STUN/TURN servers to use affect the next
gathering phase. If an ICE gathering phase has already started or
completed, the 'needs-ice-restart' bit mentioned in Section 3.5.1
will be set. This will cause the next call to createOffer to
generate new ICE credentials, for the purpose of forcing an ICE
restart and kicking off a new gathering phase, in which the new
servers will be used. If the ICE candidate pool has a nonzero
size, any existing candidates will be discarded, and new
candidates will be gathered from the new servers.
o Any change to the ICE candidate policy affects the next gathering
phase. If an ICE gathering phase has already started or
completed, the 'needs-ice-restart' bit will be set. Either way,
changes to the policy have no effect on the candidate pool,
because pooled candidates are not surfaced to the application
until a gathering phase occurs, and so any necessary filtering can
still be done on any pooled candidates.
o Any changes to the ICE candidate pool size take effect
immediately; if increased, additional candidates are pre-gathered;
if decreased, the now-superfluous candidates are discarded.
o The bundle and RTCP-multiplexing policies MUST NOT be changed
after the construction of the PeerConnection.
This call may result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity
being established.
4.1.16. addIceCandidate
The addIceCandidate method provides a remote candidate to the ICE
Agent, which, if parsed successfully, will be added to the current
and/or pending remote description according to the rules defined for
Trickle ICE. The pair of MID and ufrag is used to determine the m=
section and ICE candidate generation to which the candidate belongs.
If the MID is not present, the m= line index is used to look up the
locally generated MID (see Section 5.9), which is used in place of a
supplied MID. If these values or the candidate string are invalid,
an error is generated.
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The purpose of the ufrag is to resolve ambiguities when trickle ICE
is in progress during an ICE restart. If the ufrag is absent, the
candidate MUST be assumed to belong to the most recently applied
remote description. Connectivity checks will be sent to the new
candidate.
This method can also be used to provide an end-of-candidates
indication to the ICE Agent, as defined in [I-D.ietf-ice-trickle]).
The MID and ufrag are used as described above to determine the m=
section and ICE generation for which candidate gathering is complete.
If the ufrag is not present, then the end-of-candidates indication
MUST be assumed to apply to the relevant m= section in the most
recently applied remote description. If neither the MID nor the m=
index is present, then the indication MUST be assumed to apply to all
m= sections in the most recently applied remote description.
This call will result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity
being established.
4.2. RtpTransceiver4.2.1. stop
The stop method stops an RtpTransceiver. This will cause future
calls to createOffer to generate a zero port for the associated m=
section. See below for more details.
4.2.2. stopped
The stopped method returns "true" if the transceiver has been
stopped, either by a call to stopTransceiver or by applying an answer
that rejects the associated m= section, and "false" otherwise.
A stopped RtpTransceiver does not send any outgoing RTP or RTCP or
process any incoming RTP or RTCP. It cannot be restarted.
4.2.3. setDirection
The setDirection method sets the direction of a transceiver, which
affects the direction attribute of the associated m= section on
future calls to createOffer and createAnswer.
When creating offers, the transceiver direction is directly reflected
in the output, even for reoffers. When creating answers, the
transceiver direction is intersected with the offered direction, as
explained in the Section 5.3 section below.
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The setCodecPreferences method sets the codec preferences of a
transceiver, which in turn affect the presence and order of codecs of
the associated m= section on future calls to createOffer and
createAnswer. Note that setCodecPreferences does not directly affect
which codec the implemtation decides to send. It only affects which
codecs the implementation indicates that it prefers to receive, via
the offer or answer. Even when a codec is excluded by
setCodecPreferences, it still may be used to send until the next
offer/answer exchange discards it.
The codec preferences of an RtpTransceiver can cause codecs to be
excluded by subsequent calls to createOffer and createAnswer, in
which case the corresponding media formats in the associated m=
section will be excluded. The codec preferences cannot add media
formats that would otherwise not be present. This includes codecs
that were not negotiated in a previous offer/answer exchange that
included the transceiver.
The codec preferences of an RtpTransceiver can also determine the
order of codecs in subsequent calls to createOffer and createAnswer,
in which case the order of the media formats in the associated m=
section will match. However, the codec preferences cannot change the
order of the media formats after an answer containing the transceiver
has been applied. At this point, codecs can only be removed, not
reordered.
5. SDP Interaction Procedures
This section describes the specific procedures to be followed when
creating and parsing SDP objects.
5.1. Requirements Overview
JSEP implementations must comply with the specifications listed below
that govern the creation and processing of offers and answers.
The first set of specifications is the "mandatory-to-implement" set.
All implementations must support these behaviors, but may not use all
of them if the remote side, which may not be a JSEP endpoint, does
not support them.
The second set of specifications is the "mandatory-to-use" set. The
local JSEP endpoint and any remote endpoint must indicate support for
these specifications in their session descriptions.
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This list of mandatory-to-implement specifications is derived from
the requirements outlined in [I-D.ietf-rtcweb-rtp-usage].
R-1 [RFC4566] is the base SDP specification and MUST be
implemented.
R-2 [RFC5764] MUST be supported for signaling the UDP/TLS/RTP/SAVPF
[RFC5764], TCP/DTLS/RTP/SAVPF
[I-D.nandakumar-mmusic-proto-iana-registration], "UDP/DTLS/
SCTP" [I-D.ietf-mmusic-sctp-sdp], and "TCP/DTLS/SCTP"
[I-D.ietf-mmusic-sctp-sdp] RTP profiles.
R-3 [RFC5245] MUST be implemented for signaling the ICE credentials
and candidate lines corresponding to each media stream. The
ICE implementation MUST be a Full implementation, not a Lite
implementation.
R-4 [RFC5763] MUST be implemented to signal DTLS certificate
fingerprints.
R-5 [RFC4568] MUST NOT be implemented to signal SDES SRTP keying
information.
R-6 The [RFC5888] grouping framework MUST be implemented for
signaling grouping information, and MUST be used to identify m=
lines via the a=mid attribute.
R-7 [I-D.ietf-mmusic-msid] MUST be supported, in order to signal
associations between RTP objects and W3C MediaStreams and
MediaStreamTracks in a standard way.
R-8 The bundle mechanism in
[I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to
signal the ability to multiplex RTP streams on a single UDP
port, in order to avoid excessive use of port number resources.
R-9 The SDP attributes of "sendonly", "recvonly", "inactive", and
"sendrecv" from [RFC4566] MUST be implemented to signal
information about media direction.
R-10 [RFC5576] MUST be implemented to signal RTP SSRC values and
grouping semantics.
R-11 [RFC4585] MUST be implemented to signal RTCP based feedback.
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R-12 [RFC5761] MUST be implemented to signal multiplexing of RTP and
RTCP.
R-13 [RFC5506] MUST be implemented to signal reduced-size RTCP
messages.
R-14 [RFC4588] MUST be implemented to signal RTX payload type
associations.
R-15 [RFC3556] with bandwidth modifiers MAY be supported for
specifying RTCP bandwidth as a fraction of the media bandwidth,
RTCP fraction allocated to the senders and setting maximum
media bit-rate boundaries.
R-16 TODO: any others?
As required by [RFC4566], Section 5.13, JSEP implementations MUST
ignore unknown attribute (a=) lines.
5.1.2. Usage Requirements
All session descriptions handled by JSEP endpoints, both local and
remote, MUST indicate support for the following specifications. If
any of these are absent, this omission MUST be treated as an error.
R-1 ICE, as specified in [RFC5245], MUST be used. Note that the
remote endpoint may use a Lite implementation; implementations
MUST properly handle remote endpoints which do ICE-Lite.
R-2 DTLS [RFC6347] or DTLS-SRTP [RFC5763], MUST be used, as
appropriate for the media type, as specified in
[I-D.ietf-rtcweb-security-arch]
5.1.3. Profile Names and Interoperability
For media m= sections, JSEP endpoints MUST support both the "UDP/TLS/
RTP/SAVPF" and "TCP/DTLS/RTP/SAVPF" profiles and MUST indicate one of
these two profiles for each media m= line they produce in an offer.
For data m= sections, JSEP endpoints must support both the "UDP/DTLS/
SCTP" and "TCP/DTLS/SCTP" profiles and MUST indicate one of these two
profiles for each data m= line they produce in an offer. Because ICE
can select either TCP or UDP transport depending on network
conditions, both advertisements are consistent with ICE eventually
selecting either either UDP or TCP.
Unfortunately, in an attempt at compatibility, some endpoints
generate other profile strings even when they mean to support one of
these profiles. For instance, an endpoint might generate "RTP/AVP"
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but supply "a=fingerprint" and "a=rtcp-fb" attributes, indicating its
willingness to support "(UDP,TCP)/TLS/RTP/SAVPF". In order to
simplify compatibility with such endpoints, JSEP endpoints MUST
follow the following rules when processing the media m= sections in
an offer:
o The profile in any "m=" line in any answer MUST exactly match the
profile provided in the offer.
o Any profile matching the following patterns MUST be accepted:
"RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]"
o Because DTLS-SRTP is REQUIRED, the choice of SAVP or AVP has no
effect; support for DTLS-SRTP is determined by the presence of one
or more "a=fingerprint" attribute. Note that lack of an
"a=fingerprint" attribute will lead to negotiation failure.
o The use of AVPF or AVP simply controls the timing rules used for
RTCP feedback. If AVPF is provided, or an "a=rtcp-fb" attribute
is present, assume AVPF timing, i.e., a default value of "trr-
int=0". Otherwise, assume that AVPF is being used in an AVP
compatible mode and use AVP timing, i.e., "trr-int=4".
o For data m= sections, JSEP endpoints MUST support receiving the
"UDP/ DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards
compatibility) profiles.
Note that re-offers by JSEP endpoints MUST use the correct profile
strings even if the initial offer/answer exchange used an (incorrect)
older profile string.
5.2. Constructing an Offer
When createOffer is called, a new SDP description must be created
that includes the functionality specified in
[I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are
explained below.
5.2.1. Initial Offers
When createOffer is called for the first time, the result is known as
the initial offer.
The first step in generating an initial offer is to generate session-
level attributes, as specified in [RFC4566], Section 5.
Specifically:
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o The first SDP line MUST be "v=0", as specified in [RFC4566],
Section 5.1
o The second SDP line MUST be an "o=" line, as specified in
[RFC4566], Section 5.2. The value of the <username> field SHOULD
be "-". [RFC3264] requires that the <sess-id> be representable as
a 64-bit signed integer. It is RECOMMENDED that the <sess-id> be
generated as a 64-bit quantity with the high bit being sent to
zero and the remaining 63 bits being cryptographically random.
The value of the <nettype> <addrtype> <unicast-address> tuple
SHOULD be set to a non-meaningful address, such as IN IP4 0.0.0.0,
to prevent leaking the local address in this field. As mentioned
in [RFC4566], the entire o= line needs to be unique, but selecting
a random number for <sess-id> is sufficient to accomplish this.
o The third SDP line MUST be a "s=" line, as specified in [RFC4566],
Section 5.3; to match the "o=" line, a single dash SHOULD be used
as the session name, e.g. "s=-". Note that this differs from the
advice in [RFC4566] which proposes a single space, but as both
"o=" and "s=" are meaningless, having the same meaningless value
seems clearer.
o Session Information ("i="), URI ("u="), Email Address ("e="),
Phone Number ("p="), Bandwidth ("b="), Repeat Times ("r="), and
Time Zones ("z=") lines are not useful in this context and SHOULD
NOT be included.
o Encryption Keys ("k=") lines do not provide sufficient security
and MUST NOT be included.
o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
both <start-time> and <stop-time> SHOULD be set to zero, e.g. "t=0
0".
o An "a=ice-options" line with the "trickle" option MUST be added,
as specified in [I-D.ietf-ice-trickle], Section 4.
The next step is to generate m= sections, as specified in [RFC4566]
Section 5.14. An m= section is generated for each RtpTransceiver
that has been added to the PeerConnection. This is done in the order
that their associated RtpTransceivers were added to the
PeerConnection and excludes RtpTransceivers that are stopped and not
associated with an m= section (either due to an m= section being
recycled or an RtpTransceiver having been stopped before being
associated with an m= section) .
Each m= section, provided it is not marked as bundle-only, MUST
generate a unique set of ICE credentials and gather its own unique
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set of ICE candidates. Bundle-only m= sections MUST NOT contain any
ICE credentials and MUST NOT gather any candidates.
For DTLS, all m= sections MUST use all the certificate(s) that have
been specified for the PeerConnection; as a result, they MUST all
have the same [I-D.ietf-mmusic-4572-update] fingerprint value(s), or
these value(s) MUST be session-level attributes.
Each m= section should be generated as specified in [RFC4566],
Section 5.14. For the m= line itself, the following rules MUST be
followed:
o The port value is set to the port of the default ICE candidate for
this m= section, but given that no candidates have yet been
gathered, the "dummy" port value of 9 (Discard) MUST be used, as
indicated in [I-D.ietf-ice-trickle], Section 5.1.
o To properly indicate use of DTLS, the <proto> field MUST be set to
"UDP/TLS/RTP/SAVPF", as specified in [RFC5764], Section 8, if the
default candidate uses UDP transport, or "TCP/DTLS/RTP/SAVPF", as
specified in [I-D.nandakumar-mmusic-proto-iana-registration] if
the default candidate uses TCP transport.
o If codec preferences have been set for the associated transceiver,
media formats MUST be generated in the corresponding order, and
MUST exclude any codecs not present in the codec preferences.
o Unless excluded by the above restrictions, the media formats MUST
include the mandatory audio/video codecs as specified in
[I-D.ietf-rtcweb-audio](see Section 3) and
[I-D.ietf-rtcweb-video](see Section 5).
The m= line MUST be followed immediately by a "c=" line, as specified
in [RFC4566], Section 5.7. Again, as no candidates have yet been
gathered, the "c=" line must contain the "dummy" value "IN IP4
0.0.0.0", as defined in [I-D.ietf-ice-trickle], Section 5.1.
[I-D.ietf-mmusic-sdp-mux-attributes] groups SDP attributes into
different categories. To avoid unnecessary duplication when
bundling, Section 8.1 of [I-D.ietf-mmusic-sdp-bundle-negotiation]
specifies that attributes of category IDENTICAL or TRANSPORT should
not be repeated in bundled m= sections.
The following attributes, which are of a category other than
IDENTICAL or TRANSPORT, MUST be included in each m= section:
o An "a=mid" line, as specified in [RFC5888], Section 4. When
generating mid values, it is RECOMMENDED that the values be 3
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bytes or less, to allow them to efficiently fit into the RTP
header extension defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 11.
o A direction attribute which is the same as that of the associated
transceiver.
o For each media format on the m= line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264],
Section 5.1.
o If this m= section is for media with configurable frame sizes,
e.g. audio, an "a=maxptime" line, indicating the smallest of the
maximum supported frame sizes out of all codecs included above, as
specified in [RFC4566], Section 6.
o If this m= section is for video media, and there are known
limitations on the size of images which can be decoded, an
"a=imageattr" line, as specified in Section 3.6.
o For each primary codec where RTP retransmission should be used, a
corresponding "a=rtpmap" line indicating "rtx" with the clock rate
of the primary codec and an "a=fmtp" line that references the
payload type of the primary codec, as specified in [RFC4588],
Section 8.1.
o For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines,
as specified in [RFC4566], Section 6. The FEC mechanisms that
MUST be supported are specified in [I-D.ietf-rtcweb-fec],
Section 6, and specific usage for each media type is outlined in
Sections 4 and 5.
o For each supported RTP header extension, an "a=extmap" line, as
specified in [RFC5285], Section 5. The list of header extensions
that SHOULD/MUST be supported is specified in
[I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions
that require encryption MUST be specified as indicated in
[RFC6904], Section 4.
o For each supported RTCP feedback mechanism, an "a=rtcp-fb"
mechanism, as specified in [RFC4585], Section 4.2. The list of
RTCP feedback mechanisms that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1.
o If the bundle policy for this PeerConnection is set to "max-
bundle", and this is not the first m= section, or the bundle
policy is set to "balanced", and this is not the first m= section
for this media type, an "a=bundle-only" line.
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o If the RtpTransceiver has a sendrecv or sendonly direction:
* An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
Section 2.
o If the RtpTransceiver has a sendrecv or sendonly direction, and
the application has specified RID values or has specified more
than one encoding in the RtpSenders's parameters, an "a=rid" line
for each encoding specified. The "a=rid" line is specified in
[I-D.ietf-mmusic-rid], and its direction MUST be "send". If the
application has chosen a RID value, it MUST be used as the rid-
identifier; otherwise a RID value MUST be generated by the
implementation. When generating RID values, it is RECOMMENDED
that the values be 3 bytes or less, to allow them to efficiently
fit into the RTP header extension defined in
[I-D.ietf-avtext-rid], Section 11. If no encodings have been
specified, or only one encoding is specified but without a RID
value, then no "a=rid" lines are generated.
o If the RtpTransceiver has a sendrecv or sendonly direction and
more than one "a=rid" line has been generated, an "a=simulcast"
line, with direction "send", as defined in
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2. The list of RIDs
MUST include all of the RID identifiers used in the "a=rid" lines
for this m= section.
The following attributes, which are of category IDENTICAL or
TRANSPORT, MUST appear only in "m=" sections which either have a
unique address or which are associated with the bundle-tag. (In
initial offers, this means those "m=" sections which do not contain
an "a=bundle-only" attribute.
o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245],
Section 15.4.
o An "a=fingerprint" line for each of the endpoint's certificates,
as specified in [RFC4572], Section 5; the digest algorithm used
for the fingerprint MUST match that used in the certificate
signature.
o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the offer MUST be "actpass".
o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp]
Section 5.2.
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o An "a=rtcp" line, as specified in [RFC3605], Section 2.1,
containing the dummy value "9 IN IP4 0.0.0.0", because no
candidates have yet been gathered.
o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.1.
o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.
Lastly, if a data channel has been created, a m= section MUST be
generated for data. The <media> field MUST be set to "application"
and the <proto> field MUST be set to "UDP/DTLS/SCTP" if the default
candidate uses UDP transport, or "TCP/DTLS/SCTP" if the default
candidate uses TCP transport [I-D.ietf-mmusic-sctp-sdp]. The "fmt"
value MUST be set to "webrtc-datachannel" as specified in
[I-D.ietf-mmusic-sctp-sdp], Section 4.1.
Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-pwd",
"a=fingerprint", "a=dtls-id", and "a=setup" lines MUST be included as
mentioned above, along with an "a=fmtp:webrtc-datachannel" line and
an "a=sctp-port" line referencing the SCTP port number as defined in
[I-D.ietf-mmusic-sctp-sdp], Section 4.1.
Once all m= sections have been generated, a session-level "a=group"
attribute MUST be added as specified in [RFC5888]. This attribute
MUST have semantics "bundle", and MUST include the mid identifiers of
each m= section. The effect of this is that the browser offers all
m= sections as one bundle group. However, whether the m= sections
are bundle-only or not depends on the bundle policy.
The next step is to generate session-level lip sync groups as defined
in [RFC5888], Section 7. For each MediaStream referenced by more
than one RtpTransceiver (by passing those MediaStreams as arguments
to the addTrack and addTransceiver methods), a group of type "LS"
MUST be added that contains the mid values for each RtpTransceiver.
Attributes which SDP permits to either be at the session level or the
media level SHOULD generally be at the media level even if they are
identical. This promotes readability, especially if one of a set of
initially identical attributes is subsequently changed.
Attributes other than the ones specified above MAY be included,
except for the following attributes which are specifically
incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
and MUST NOT be included:
o "a=crypto"
o "a=key-mgmt"
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o "a=ice-lite"
Note that when bundle is used, any additional attributes that are
added MUST follow the advice in [I-D.ietf-mmusic-sdp-mux-attributes]
on how those attributes interact with bundle.
Note that these requirements are in some cases stricter than those of
SDP. Implementations MUST be prepared to accept compliant SDP even
if it would not conform to the requirements for generating SDP in
this specification.
5.2.2. Subsequent Offers
When createOffer is called a second (or later) time, or is called
after a local description has already been installed, the processing
is somewhat different than for an initial offer.
If the initial offer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "stable" state, the steps
for generating an initial offer should be followed, subject to the
following restriction:
o The fields of the "o=" line MUST stay the same except for the
<session-version> field, which MUST increment by one on each call
to createOffer if the offer might differ from the output of the
previous call to createOffer; implementations MAY opt to increment
<session-version> on every call. The value of the generated
<session-version> is independent of the <session-version> of the
current local description; in particular, in the case where the
current version is N, an offer is created with version N+1, and
then that offer is rolled back so that the current version is
again N, the next generated offer will still have version N+2.
Note that if the application creates an offer by reading
currentLocalDescription instead of calling createOffer, the returned
SDP may be different than when setLocalDescription was originally
called, due to the addition of gathered ICE candidates, but the
<session-version> will not have changed. There are no known
scenarios in which this causes problems, but if this is a concern,
the solution is simply to use createOffer to ensure a unique
<session-version>.
If the initial offer was applied using setLocalDescription, but an
answer from the remote side has not yet been applied, meaning the
PeerConnection is still in the "local-offer" state, an offer is
generated by following the steps in the "stable" state above, along
with these exceptions:
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o The "s=" and "t=" lines MUST stay the same.
o If any RtpTransceiver has been added, and there exists an m=
section with a zero port in the current local description or the
current remote description, that m= section MUST be recycled by
generating an m= section for the added RtpTransceiver as if the m=
section were being added to the session description, placed at the
same index as the m= section with a zero port.
o If an RtpTransceiver is stopped and is not associated with an m=
section, an m= section MUST NOT be generated for it. This
prevents adding back RtpTransceivers whose m= sections were
recycled and used for a new RtpTransceiver in a previous offer/
answer exchange, as described above.
o If an RtpTransceiver has been stopped and is associated with an m=
section, and the m= section is not being recycled as described
above, an m= section MUST be generated for it with the port set to
zero and the "a=msid" line removed.
o For RtpTransceivers that are not stopped, the "a=msid" line MUST
stay the same if they are present in the current description.
o Each "m=" and c=" line MUST be filled in with the port, protocol,
and address of the default candidate for the m= section, as
described in [RFC5245], Section 4.3. If ICE checking has already
completed for one or more candidate pairs and a candidate pair is
in active use, then that pair MUST be used, even if ICE has not
yet completed. Note that this differs from the guidance in
[RFC5245], Section 9.1.2.2, which only refers to offers created
when ICE has completed. In each case, if no RTP candidates have
yet been gathered, dummy values MUST be used, as described above.
o Each "a=mid" line MUST stay the same.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
the ICE configuration has changed (either changes to the supported
STUN/TURN servers, or the ICE candidate policy), or the
"IceRestart" option ( Section 5.2.3.1 was specified. If the m=
section is bundled into another m= section, it still MUST NOT
contain any ICE credentials.
o If the m= section is not bundled into another m= section, an
"a=rtcp" attribute line MUST be added with of the default RTCP
candidate, as indicated in [RFC5761], section 5.1.3.
o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering
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phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate
gathering for the section has completed, an "a=end-of-candidates"
attribute MUST be added, as described in [I-D.ietf-ice-trickle],
Section 9.3. If the m= section is bundled into another m=
section, both "a=candidate" and "a=end-of-candidates" MUST be
omitted.
o For RtpTransceivers that are still present, the "a=msid" line MUST
stay the same.
o For RtpTransceivers that are still present, the "a=rid" lines MUST
stay the same.
o For RtpTransceivers that are still present, any "a=simulcast" line
MUST stay the same.
o If any RtpTransceiver has been stopped, the port MUST be set to
zero and the "a=msid" line MUST be removed.
o If any RtpTransceiver has been added, and there exists a m=
section with a zero port in the current local description or the
current remote description, that m= section MUST be recycled by
generating a m= section for the added RtpTransceiver as if the m=
section were being added to session description, except that
instead of adding it, the generated m= section replaces the m=
section with a zero port.
If the initial offer was applied using setLocalDescription, and an
answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "remote-
pranswer" or "stable" states, an offer is generated based on the
negotiated session descriptions by following the steps mentioned for
the "local-offer" state above.
In addition, for each non-recycled, non-rejected m= section in the
new offer, the following adjustments are made based on the contents
of the corresponding m= section in the current remote description:
o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include codecs present in the most recent answer which have
not been excluded by the codec preferences of the associated
transceiver.
o The media formats on the m= line MUST be generated in the same
order as in the current local description.
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o The RTP header extensions MUST only include those that are present
in the most recent answer.
o The RTCP feedback extensions MUST only include those that are
present in the most recent answer.
o The "a=rtcp" line MUST only be added if the most recent answer did
not include an "a=rtcp-mux" line.
o The "a=rtcp-mux" line MUST only be added if present in the most
recent answer.
o The "a=rtcp-mux-only" line MUST only be added if present in the
most recent answer.
o The "a=rtcp-rsize" line MUST only be added if present in the most
recent answer.
The "a=group:BUNDLE" attribute MUST include the mid identifiers
specified in the bundle group in the most recent answer, minus any m=
sections that have been marked as rejected, plus any newly added or
re-enabled m= sections. In other words, the bundle attribute must
contain all m= sections that were previously bundled, as long as they
are still alive, as well as any new m= sections.
The "LS" groups are generated in the same way as with initial offers.
5.2.3. Options Handling
The createOffer method takes as a parameter an RTCOfferOptions
object. Special processing is performed when generating a SDP
description if the following options are present.
5.2.3.1. IceRestart
If the "IceRestart" option is specified, with a value of "true", the
offer MUST indicate an ICE restart by generating new ICE ufrag and
pwd attributes, as specified in [RFC5245], Section 9.1.1.1. If this
option is specified on an initial offer, it has no effect (since a
new ICE ufrag and pwd are already generated). Similarly, if the ICE
configuration has changed, this option has no effect, since new ufrag
and pwd attributes will be generated automatically. This option is
primarily useful for reestablishing connectivity in cases where
failures are detected by the application.
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If the "VoiceActivityDetection" option is specified, with a value of
"true", the offer MUST indicate support for silence suppression in
the audio it receives by including comfort noise ("CN") codecs for
each offered audio codec, as specified in [RFC3389], Section 5.1,
except for codecs that have their own internal silence suppression
support. For codecs that have their own internal silence suppression
support, the appropriate fmtp parameters for that codec MUST be
specified to indicate that silence suppression for received audio is
desired. For example, when using the Opus codec, the "usedtx=1"
parameter would be specified in the offer. This option allows the
endpoint to significantly reduce the amount of audio bandwidth it
receives, at the cost of some fidelity, depending on the quality of
the remote VAD algorithm.
If the "VoiceActivityDetection" option is specified, with a value of
"false", the browser MUST NOT emit "CN" codecs. For codecs that have
their own internal silence suppression support, the appropriate fmtp
parameters for that codec MUST be specified to indicate that silence
suppression for received audio is not desired. For example, when
using the Opus codec, the "usedtx=0" parameter would be specified in
the offer.
Note that setting the "VoiceActivityDetection" parameter when
generating an offer is a request to receive audio with silence
suppression. It has no impact on whether the local endpoint does
silence suppression for the audio it sends.
The "VoiceActivityDetection" option does not have any impact on the
setting of the "vad" value in the signaling of the client to mixer
audio level header extension described in [RFC6464], Section 4.
5.3. Generating an Answer
When createAnswer is called, a new SDP description must be created
that is compatible with the supplied remote description as well as
the requirements specified in [I-D.ietf-rtcweb-rtp-usage]. The exact
details of this process are explained below.
5.3.1. Initial Answers
When createAnswer is called for the first time after a remote
description has been provided, the result is known as the initial
answer. If no remote description has been installed, an answer
cannot be generated, and an error MUST be returned.
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Note that the remote description SDP may not have been created by a
JSEP endpoint and may not conform to all the requirements listed in
Section 5.2. For many cases, this is not a problem. However, if any
mandatory SDP attributes are missing, or functionality listed as
mandatory-to-use above is not present, this MUST be treated as an
error, and MUST cause the affected m= sections to be marked as
rejected.
The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that
indicated in the Initial Offers section above, except that the
"a=ice-options" line, with the "trickle" option as specified in
[I-D.ietf-ice-trickle], Section 4, is only included if such an option
was present in the offer.
The next step is to generate session-level lip sync groups as defined
in [RFC5888], Section 7. For each group of type "LS" present in the
offer, determine which of the local RtpTransceivers identified by
that group's mid values reference a common local MediaStream (as
specified in the addTrack and addTransceiver methods). If at least
two such RtpTransceivers exist, a group of type "LS" with the mid
values of these RtpTransceivers MUST be added. Otherwise, this
indicates a difference of opinion between the offerer and answerer
regarding lip sync status, and as such, the offered group MUST be
ignored and no corresponding "LS" group generated.
The next step is to generate m= sections for each m= section that is
present in the remote offer, as specified in [RFC3264], Section 6.
For the purposes of this discussion, any session-level attributes in
the offer that are also valid as media-level attributes SHALL be
considered to be present in each m= section.
The next step is to go through each offered m= section. Each offered
m= section will have an associated RtpTransceiver, as described in
Section 5.9. If there are more RtpTransceivers than there are m=
sections, the unmatched RtpTransceivers will need to be associated in
a subsequent offer.
For each offered m= section, if any of the following conditions are
true, the corresponding m= section in the answer MUST be marked as
rejected by setting the port in the m= line to zero, as indicated in
[RFC3264], Section 6., and further processing for this m= section can
be skipped:
o The associated RtpTransceiver has been stopped.
o No supported codec is present in the offer.
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o The bundle policy is "max-bundle", and this is not the first m=
section or in the same bundle group as the first m= section.
o The bundle policy is "balanced", and this is not the first m=
section for this media type or in the same bundle group as the
first m= section for this media type.
o The RTP/RTCP multiplexing policy is "require" and the m= section
doesn't contain an "a=rtcp-mux" attribute.
Otherwise, each m= section in the answer should then be generated as
specified in [RFC3264], Section 6.1. For the m= line itself, the
following rules must be followed:
o The port value would normally be set to the port of the default
ICE candidate for this m= section, but given that no candidates
have yet been gathered, the "dummy" port value of 9 (Discard) MUST
be used, as indicated in [I-D.ietf-ice-trickle], Section 5.1.
o The <proto> field MUST be set to exactly match the <proto> field
for the corresponding m= line in the offer.
o If codec preferences have been set for the associated transceiver,
media formats MUST be generated in the corresponding order, and
MUST exclude any codecs not present in the codec preferences or
not present in the offer.
o Unless excluded by the above restrictions, the media formats MUST
include the mandatory audio/video codecs as specified in
[I-D.ietf-rtcweb-audio](see Section 3) and
[I-D.ietf-rtcweb-video](see Section 5).
The m= line MUST be followed immediately by a "c=" line, as specified
in [RFC4566], Section 5.7. Again, as no candidates have yet been
gathered, the "c=" line must contain the "dummy" value "IN IP4
0.0.0.0", as defined in [I-D.ietf-ice-trickle], Section 5.1.
If the offer supports bundle, all m= sections to be bundled must use
the same ICE credentials and candidates; all m= sections not being
bundled must use unique ICE credentials and candidates. Each m=
section MUST contain the following attributes (which are of attribute
types other than IDENTICAL and TRANSPORT):
o If and only if present in the offer, an "a=mid" line, as specified
in [RFC5888], Section 9.1. The "mid" value MUST match that
specified in the offer.
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o A direction attribute, determined by applying the rules regarding
the offered direction specified in [RFC3264], Section 6.1, and
then intersecting with the direction of the associated
RtpTransceiver. For example, in the case where an m= section is
offered as "sendonly", and the local transceiver is set to
"sendrecv", the result in the answer is a "recvonly" direction.
o For each media format on the m= line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264],
Section 6.1.
o If this m= section is for media with configurable frame sizes,
e.g. audio, an "a=maxptime" line, indicating the smallest of the
maximum supported frame sizes out of all codecs included above, as
specified in [RFC4566], Section 6.
o If this m= section is for video media, and there are known
limitations on the size of images which can be decoded, an
"a=imageattr" line, as specified in Section 3.6.
o If "rtx" is present in the offer, for each primary codec where RTP
retransmission should be used, a corresponding "a=rtpmap" line
indicating "rtx" with the clock rate of the primary codec and an
"a=fmtp" line that references the payload type of the primary
codec, as specified in [RFC4588], Section 8.1.
o For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines,
as specified in [RFC4566], Section 6. The FEC mechanisms that
MUST be supported are specified in [I-D.ietf-rtcweb-fec],
Section 6, and specific usage for each media type is outlined in
Sections 4 and 5.
o For each supported RTP header extension that is present in the
offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
The list of header extensions that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header
extensions that require encryption MUST be specified as indicated
in [RFC6904], Section 4.
o For each supported RTCP feedback mechanism that is present in the
offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585],
Section 4.2. The list of RTCP feedback mechanisms that SHOULD/
MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage],
Section 5.1.
o If the RtpTransceiver has a sendrecv or sendonly direction:
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* An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
Section 2.
Each m= section which is not bundled into another m= section, MUST
contain the following attributes (which are of category IDENTICAL or
TRANSPORT):
o "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245],
Section 15.4.
o An "a=fingerprint" line for each of the endpoint's certificates,
as specified in [RFC4572], Section 5; the digest algorithm used
for the fingerprint MUST match that used in the certificate
signature.
o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the answer MUST be "active" or "passive"; the
"active" role is RECOMMENDED. The role value MUST be consistent
with the existing DTLS connection, if one exists and is being
continued.
o An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp]
Section 5.3.
o If present in the offer, an "a=rtcp-mux" line, as specified in
[RFC5761], Section 5.1.1. Otherwise, an "a=rtcp" line, as
specified in [RFC3605], Section 2.1, containing the dummy value "9
IN IP4 0.0.0.0" (because no candidates have yet been gathered).
o If present in the offer, an "a=rtcp-rsize" line, as specified in
[RFC5506], Section 5.
If a data channel m= section has been offered, a m= section MUST also
be generated for data. The <media> field MUST be set to
"application" and the <proto> and "fmt" fields MUST be set to exactly
match the fields in the offer.
Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-pwd",
"a=candidate", "a=fingerprint", "a=dtls-id", and "a=setup" lines MUST
be included under the conditions described above, along with an
"a=fmtp:webrtc-datachannel" line and an "a=sctp-port" line
referencing the SCTP port number as defined in
[I-D.ietf-mmusic-sctp-sdp], Section 4.1.
If "a=group" attributes with semantics of "BUNDLE" are offered,
corresponding session-level "a=group" attributes MUST be added as
specified in [RFC5888]. These attributes MUST have semantics
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"BUNDLE", and MUST include the all mid identifiers from the offered
bundle groups that have not been rejected. Note that regardless of
the presence of "a=bundle-only" in the offer, no m= sections in the
answer should have an "a=bundle-only" line.
Attributes that are common between all m= sections MAY be moved to
session-level, if explicitly defined to be valid at session-level.
The attributes prohibited in the creation of offers are also
prohibited in the creation of answers.
5.3.2. Subsequent Answers
When createAnswer is called a second (or later) time, or is called
after a local description has already been installed, the processing
is somewhat different than for an initial answer.
If the initial answer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "have-remote-offer" state,
the steps for generating an initial answer should be followed,
subject to the following restriction:
o The fields of the "o=" line MUST stay the same except for the
<session-version> field, which MUST increment if the session
description changes in any way from the previously generated
answer.
If any session description was previously supplied to
setLocalDescription, an answer is generated by following the steps in
the "have-remote-offer" state above, along with these exceptions:
o The "s=" and "t=" lines MUST stay the same.
o Each "m=" and c=" line MUST be filled in with the port and address
of the default candidate for the m= section, as described in
[RFC5245], Section 4.3. Note, however, that the m= line protocol
need not match the default candidate, because this protocol value
must instead match what was supplied in the offer, as described
above.
o The media formats on the m= line MUST be generated in the same
order as in the current local description.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
the m= section is restarting, in which case new ICE credentials
must be created as specified in [RFC5245], Section 9.2.1.1. If
the m= section is bundled into another m= section, it still MUST
NOT contain any ICE credentials.
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o If the m= section is not bundled into another m= section and RTCP
multiplexing is not active, an "a=rtcp" attribute line MUST be
filled in with the port and address of the default RTCP candidate.
If no RTCP candidates have yet been gathered, dummy values MUST be
used, as described in the initial answer section above.
o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate
gathering for the section has completed, an "a=end-of-candidates"
attribute MUST be added, as described in [I-D.ietf-ice-trickle],
Section 9.3. If the m= section is bundled into another m=
section, both "a=candidate" and "a=end-of-candidates" MUST be
omitted.
o For RtpTransceivers that are not stopped, the "a=msid" line MUST
stay the same.
5.3.3. Options Handling
The createAnswer method takes as a parameter an RTCAnswerOptions
object. The set of parameters for RTCAnswerOptions is different than
those supported in RTCOfferOptions; the IceRestart option is
unnecessary, as ICE credentials will automatically be changed for all
m= lines where the offerer chose to perform ICE restart.
The following options are supported in RTCAnswerOptions.
5.3.3.1. VoiceActivityDetection
Silence suppression in the answer is handled as described in
Section 5.2.3.2, with one exception: if support for silence
suppression was not indicated in the offer, the
VoiceActivityDetection parameter has no effect, and the answer should
be generated as if VoiceActivityDetection was set to false. This is
done on a per-codec basis (e.g., if the offerer somehow offered
support for CN but set "usedtx=0" for Opus, setting
VoiceActivityDetection to true would result in an answer with CN
codecs and "usedtx=0").
5.4. Modifying an Offer or Answer
The SDP returned from createOffer or createAnswer MUST NOT be changed
before passing it to setLocalDescription. If precise control over
the SDP is needed, the aformentioned createOffer/createAnswer options
or RTPSender APIs MUST be used.
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Note that the application MAY modify the SDP to reduce the
capabilities in the offer it sends to the far side (post-
setLocalDescription) or the offer that it installs from the far side
(pre-setRemoteDescription), as long as it remains a valid SDP offer
and specifies a subset of what was in the original offer. This is
safe because the answer is not permitted to expand capabilities, and
therefore will just respond to what is present in the offer.
The application SHOULD NOT modify the SDP in the answer it transmits,
as the answer contains the negotiated capabilities, and this can
cause the two sides to have different ideas about what exactly was
negotiated.
As always, the application is solely responsible for what it sends to
the other party, and all incoming SDP will be processed by the
browser to the extent of its capabilities. It is an error to assume
that all SDP is well-formed; however, one should be able to assume
that any implementation of this specification will be able to
process, as a remote offer or answer, unmodified SDP coming from any
other implementation of this specification.
5.5. Processing a Local Description
When a SessionDescription is supplied to setLocalDescription, the
following steps MUST be performed:
o First, the type of the SessionDescription is checked against the
current state of the PeerConnection:
* If the type is "offer", the PeerConnection state MUST be either
"stable" or "have-local-offer".
* If the type is "pranswer" or "answer", the PeerConnection state
MUST be either "have-remote-offer" or "have-local-pranswer".
o If the type is not correct for the current state, processing MUST
stop and an error MUST be returned.
o Next, the SessionDescription is parsed into a data structure, as
described in the Section 5.7 section below. If parsing fails for
any reason, processing MUST stop and an error MUST be returned.
o Finally, the parsed SessionDescription is applied as described in
the Section 5.8 section below.
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When a SessionDescription is supplied to setRemoteDescription, the
following steps MUST be performed:
o First, the type of the SessionDescription is checked against the
current state of the PeerConnection:
* If the type is "offer", the PeerConnection state MUST be either
"stable" or "have-remote-offer".
* If the type is "pranswer" or "answer", the PeerConnection state
MUST be either "have-local-offer" or "have-remote-pranswer".
o If the type is not correct for the current state, processing MUST
stop and an error MUST be returned.
o Next, the SessionDescription is parsed into a data structure, as
described in the Section 5.7 section below. If parsing fails for
any reason, processing MUST stop and an error MUST be returned.
o Finally, the parsed SessionDescription is applied as described in
the Section 5.9 section below.
5.7. Parsing a Session Description
When a SessionDescription of any type is supplied to setLocal/
RemoteDescription, the implementation must parse it and reject it if
it is invalid. The exact details of this process are explained
below.
The SDP contained in the session description object consists of a
sequence of text lines, each containing a key-value expression, as
described in [RFC4566], Section 5. The SDP is read, line-by-line,
and converted to a data structure that contains the deserialized
information. However, SDP allows many types of lines, not all of
which are relevant to JSEP applications. For each line, the
implementation will first ensure it is syntactically correct
according to its defining ABNF, check that it conforms to [RFC4566]
and [RFC3264] semantics, and then either parse and store or discard
the provided value, as described below.
If any line is not well-formed, or cannot be parsed as described, the
parser MUST stop with an error and reject the session description,
even if the value is to be discarded. This ensures that
implementations do not accidentally misinterpret ambiguous SDP.
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First, the session-level lines are checked and parsed. These lines
MUST occur in a specific order, and with a specific syntax, as
defined in [RFC4566], Section 5. Note that while the specific line
types (e.g. "v=", "c=") MUST occur in the defined order, lines of the
same type (typically "a=") can occur in any order, and their ordering
is not meaningful.
The following non-attribute lines are not meaningful in the JSEP
context and MAY be discarded once they have been checked.
The "c=" line MUST be checked for syntax but its value is not
used. This supersedes the guidance in [RFC5245], Section 6.1, to
use "ice-mismatch" to indicate mismatches between "c=" and the
candidate lines; because JSEP always uses ICE, "ice-mismatch" is
not useful in this context.
The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines are
not used by this specification; they MUST be checked for syntax
but their values are not used.
The remaining non-attribute lines are processed as follows:
The "v=" line MUST have a version of 0, as specified in [RFC4566],
Section 5.1.
The "o=" line MUST be parsed as specified in [RFC4566],
Section 5.2.
The "b=" line, if present, MUST be parsed as specified in
[RFC4566], Section 5.8, and the bwtype and bandwidth values
stored.
Finally, the attribute lines are processed. Specific processing MUST
be applied for the following session-level attribute ("a=") lines:
o Any "a=group" lines are parsed as specified in [RFC5888],
Section 5, and the group's semantics and mids are stored.
o If present, a single "a=ice-lite" line is parsed as specified in
[RFC5245], Section 15.3, and a value indicating the presence of
ice-lite is stored.
o If present, a single "a=ice-ufrag" line is parsed as specified in
[RFC5245], Section 15.4, and the ufrag value is stored.
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o If present, a single "a=ice-pwd" line is parsed as specified in
[RFC5245], Section 15.4, and the password value is stored.
o If present, a single "a=ice-options" line is parsed as specified
in [RFC5245], Section 15.5, and the set of specified options is
stored.
o Any "a=fingerprint" lines are parsed as specified in [RFC4572],
Section 5, and the set of fingerprint and algorithm values is
stored.
o If present, a single "a=setup" line is parsed as specified in
[RFC4145], Section 4, and the setup value is stored.
o If present, a single "a=dtls-id" line is parsed as specified in
[I-D.ietf-mmusic-dtls-sdp] Section 5, and the dtls-id value is
stored.
o Any "a=extmap" lines are parsed as specified in [RFC5285],
Section 5, and their values are stored.
Once all the session-level lines have been parsed, processing
continues with the lines in media sections.
5.7.2. Media Section Parsing
Like the session-level lines, the media session lines MUST occur in
the specific order and with the specific syntax defined in [RFC4566],
Section 5.
The "m=" line itself MUST be parsed as described in [RFC4566],
Section 5.14, and the media, port, proto, and fmt values stored.
Following the "m=" line, specific processing MUST be applied for the
following non-attribute lines:
o As with the "c=" line at the session level, the "c=" line MUST be
parsed according to [RFC4566], Section 5.7, but its value is not
used.
o The "b=" line, if present, MUST be parsed as specified in
[RFC4566], Section 5.8, and the bwtype and bandwidth values
stored.
Specific processing MUST also be applied for the following attribute
lines:
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o If present, a single "a=ice-ufrag" line is parsed as specified in
[RFC5245], Section 15.4, and the ufrag value is stored.
o If present, a single "a=ice-pwd" line is parsed as specified in
[RFC5245], Section 15.4, and the password value is stored.
o If present, a single "a=ice-options" line is parsed as specified
in [RFC5245], Section 15.5, and the set of specified options is
stored.
o Any "a=candidate" attributes MUST be parsed as specified in
[RFC5245], Section 15.1, and their values stored.
o Any "a=remote-candidates" attributes MUST be parsed as specified
in [RFC5245], Section 15.2, but their values are ignored.
o If present, a single "a=end-of-candidates" attribute MUST be
parsed as specified in [I-D.ietf-ice-trickle], Section 8.2, and
its presence or absence flagged and stored.
o Any "a=fingerprint" lines are parsed as specified in [RFC4572],
Section 5, and the set of fingerprint and algorithm values is
stored.
If the "m=" proto value indicates use of RTP, as described in the
Section 5.1.3 section above, the following attribute lines MUST be
processed:
o The "m=" fmt value MUST be parsed as specified in [RFC4566],
Section 5.14, and the individual values stored.
o Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as specified in
[RFC4566], Section 6, and their values stored.
o If present, a single "a=ptime" line MUST be parsed as described in
[RFC4566], Section 6, and its value stored.
o If present, a single "a=maxptime" line MUST be parsed as described
in [RFC4566], Section 6, and its value stored.
o If present, a single direction attribute line (e.g. "a=sendrecv")
MUST be parsed as described in [RFC4566], Section 6, and its value
stored.
o Any "a=ssrc" or "a=ssrc-group" attributes MUST be parsed as
specified in [RFC5576], Sections 4.1-4.2, and their values stored.
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o Any "a=extmap" attributes MUST be parsed as specified in
[RFC5285], Section 5, and their values stored.
o Any "a=rtcp-fb" attributes MUST be parsed as specified in
[RFC4585], Section 4.2., and their values stored.
o If present, a single "a=rtcp-mux" attribute MUST be parsed as
specified in [RFC5761], Section 5.1.1, and its presence or absence
flagged and stored.
o If present, a single "a=rtcp-mux-only" attribute MUST be parsed as
specified in [I-D.ietf-mmusic-mux-exclusive], Section 3, and its
presence or absence flagged and stored.
o If present, a single "a=rtcp-rsize" attribute MUST be parsed as
specified in [RFC5506], Section 5, and its presence or absence
flagged and stored.
o If present, a single "a=rtcp" attribute MUST be parsed as
specified in [RFC3605], Section 2.1, but its value is ignored, as
this information is superfluous when using ICE.
o If present, a single "a=msid" attribute MUST be parsed as
specified in [I-D.ietf-mmusic-msid], Section 3.2, and its value
stored.
o Any "a=imageattr" attributes MUST be parsed as specified in
[RFC6236], Section 3, and their values stored.
o Any "a=rid" lines MUST be parsed as specified in
[I-D.ietf-mmusic-rid], Section 10, and their values stored.
o If present, a single "a=simulcast" line MUST be parsed as
specified in [I-D.ietf-mmusic-sdp-simulcast], and its values
stored.
Otherwise, if the "m=" proto value indicates use of SCTP, the
following attribute lines MUST be processed:
o The "m=" fmt value MUST be parsed as specified in
[I-D.ietf-mmusic-sctp-sdp], Section 4.3, and the application
protocol value stored.
o An "a=sctp-port" attribute MUST be present, and it MUST be parsed
as specified in [I-D.ietf-mmusic-sctp-sdp], Section 5.2, and the
value stored.
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o If present, a single "a=max-message-size" attribute MUST be parsed
as specified in [I-D.ietf-mmusic-sctp-sdp], Section 6, and the
value stored. Otherwise, use the specified default.
5.7.3. Semantics Verification
Assuming parsing completes successfully, the parsed description is
then evaluated to ensure internal consistency as well as proper
support for mandatory features. Specifically, the following checks
are performed:
o For each m= section, valid values for each of the mandatory-to-use
features enumerated in Section 5.1.2 MUST be present. These
values MAY either be present at the media level, or inherited from
the session level.
* ICE ufrag and password values, which MUST comply with the size
limits specified in [RFC5245], Section 15.4.
* dtls-id value, which MUST be set according to
[I-D.ietf-mmusic-dtls-sdp] Section 5. If this is a re-offer
and the dtls-id value is different from that presently in use,
the DTLS connection is not being continued and the remote
description MUST be part of an ICE restart, together with new
ufrag and password values. If this is an answer, the dtls-id
value, if present, MUST be the same as in the offer.
* DTLS setup value, which MUST be set according to the rules
specified in [RFC5763], Section 5 and MUST be consistent with
the selected role of the current DTLS connection, if one exists
and is being continued.
* DTLS fingerprint values, where at least one fingerprint MUST be
present.
o All RID values referenced in an "a=simulcast" line MUST exist as
"a=rid" lines.
o Each m= section is also checked to ensure prohibited features are
not used. If this is a local description, the "ice-lite"
attribute MUST NOT be specified.
If this session description is of type "pranswer" or "answer", the
following additional checks are applied:
o The session description must follow the rules defined in
[RFC3264], Section 6, including the requirement that the number of
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m= sections MUST exactly match the number of m= sections in the
associated offer.
o For each m= section, the media type and protocol values MUST
exactly match the media type and protocol values in the
corresponding m= section in the associated offer.
5.8. Applying a Local Description
The following steps are performed at the media engine level to apply
a local description.
First, the parsed parameters are checked to ensure that they have not
been altered after their generation in createOffer/createAnswer, as
discussed in Section 5.4; otherwise, processing MUST stop and an
error MUST be returned.
Next, media sections are processed. For each media section, the
following steps MUST be performed; if any parameters are out of
bounds, or cannot be applied, processing MUST stop and an error MUST
be returned.
o If this media section is new, begin gathering candidates for it,
as defined in [RFC5245], Section 4.1.1, unless it has been marked
as bundle-only.
o Or, if the ICE ufrag and password values have changed, and it has
not been marked as bundle-only, trigger the ICE Agent to start an
ICE restart, and begin gathering new candidates for the media
section as described in [RFC5245], Section 9.1.1.1. If this
description is an answer, also start checks on that media section
as defined in [RFC5245], Section 9.3.1.1.
o If the media section proto value indicates use of RTP:
* If there is no RtpTransceiver associated with this m= section
(which should only happen when applying an offer), find one and
associate it with this m= section according to the following
steps:
+ Find the RtpTransceiver that corresponds to the m= section
with the same MID in the created offer.
+ Set the value of the RtpTransceiver's mid attribute to the
MID of the m= section.
* If RTCP mux is indicated, prepare to demux RTP and RTCP from
the RTP ICE component, as specified in [RFC5761],
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in a previous description, this MUST result in an error.
* For each specified RTP header extension, establish a mapping
between the extension ID and URI, as described in section 6 of
[RFC5285]. If any indicated RTP header extension is not
supported, this MUST result in an error.
* If the MID header extension is supported, prepare to demux RTP
data intended for this media section based on the MID header
extension, as described in [I-D.ietf-mmusic-msid], Section 3.2.
* For each specified media format, establish a mapping between
the payload type and the actual media format, as described in
[RFC3264], Section 6.1. If any indicated media format is not
supported, this MUST result in an error.
* For each specified "rtx" media format, establish a mapping
between the RTX payload type and its associated primary payload
type, as described in [RFC4588], Sections 8.6 and 8.7. If any
referenced primary payload types are not present, this MUST
result in an error.
* If the directional attribute is of type "sendrecv" or
"recvonly", enable receipt and decoding of media.
Finally, if this description is of type "pranswer" or "answer",
follow the processing defined in the Section 5.10 section below.
5.9. Applying a Remote Description
If the answer contains any "a=ice-options" attributes where "trickle"
is listed as an attribute, update the PeerConnection canTrickle
property to be true. Otherwise, set this property to false.
The following steps are performed at the media engine level to apply
a remote description.
The following steps MUST be performed for attributes at the session
level; if any parameters are out of bounds, or cannot be applied,
processing MUST stop and an error MUST be returned.
o For any specified "CT" bandwidth value, set this as the limit for
the maximum total bitrate for all m= sections, as specified in
Section 5.8 of [RFC4566]. The implementation can decide how to
allocate the available bandwidth between m= sections to
simultaneously meet any limits on individual m= sections, as well
as this overall session limit.
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o For any specified "RR" or "RS" bandwidth values, handle as
specified in [RFC3556], Section 2.
o Any "AS" bandwidth value MUST be ignored, as the meaning of this
construct at the session level is not well defined.
For each media section, the following steps MUST be performed; if any
parameters are out of bounds, or cannot be applied, processing MUST
stop and an error MUST be returned.
o If the ICE ufrag or password changed from the previous remote
description, then an ICE restart is needed, as described in
Section 9.1.1.1 of [RFC5245] If the description is of type
"offer", mark that an ICE restart is needed. If the description
is of type "answer" and the current local description is also an
ICE restart, then signal the ICE agent to begin checks as
described in Section 9.3.1.1 of [RFC5245]. An answer MUST change
the ufrag and password in an answer if and only if ICE is
restarting, as described in Section 9.2.1.1 of [RFC5245].
o Configure the ICE components associated with this media section to
use the supplied ICE remote ufrag and password for their
connectivity checks.
o Pair any supplied ICE candidates with any gathered local
candidates, as described in Section 5.7 of [RFC5245] and start
connectivity checks with the appropriate credentials.
o If an "a=end-of-candidates" attribute is present, process the end-
of-candidates indication as described in [I-D.ietf-ice-trickle]
Section 11.
o If the media section proto value indicates use of RTP:
* If the m= section is being recycled (see Section 5.2.2),
dissociate the currently associated RtpTransceiver by setting
its mid attribute to null.
* If the m= section is not associated with any RtpTransceiver
(possibly because it was dissociated in the previous step),
either find an RtpTransceiver or create one according to the
following steps:
+ If the m= section is sendrecv or recvonly, and there are
RtpTransceivers of the same type that were added to the
PeerConnection by addTrack and are not associated with any
m= section and are not stopped, find the first (according to
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the canonical order described in Section 5.2.1) such
RtpTransceiver.
+ If no RtpTransceiver was found in the previous step, create
one with a recvonly direction.
+ Associate the found or created RtpTransceiver with the m=
section by setting the value of the RtpTransceiver's mid
attribute to the MID of the m= section. If the m= section
does not include a MID (i.e., the remote side does not
support the MID extension), generate a value for the
RtpTransceiver mid attribute, following the guidance for
"a=mid" mentioned in Section 5.2.1.
* For each specified media format that is also supported by the
local implementation, establish a mapping between the specified
payload type and the media format, as described in [RFC3264],
Section 6.1. Specifically, this means that the implementation
records the payload type to be used in outgoing RTP packets
when sending each specified media format, as well as the
relative preference for each format that is indicated in their
ordering. If any indicated media format is not supported by
the local implementation, it MUST be ignored.
* For each specified "rtx" media format, establish a mapping
between the RTX payload type and its associated primary payload
type, as described in [RFC4588], Section 4. If any referenced
primary payload types are not present, this MUST result in an
error.
* For each specified fmtp parameter that is supported by the
local implementation, enable them on the associated media
formats.
* For each specified RTP header extension that is also supported
by the local implementation, establish a mapping between the
extension ID and URI, as described in [RFC5285], Section 5.
Specifically, this means that the implementation records the
extension ID to be used in outgoing RTP packets when sending
each specified header extension. If any indicated RTP header
extension is not supported by the local implementation, it MUST
be ignored.
* For each specified RTCP feedback mechanism that is supported by
the local implementation, enable them on the associated media
formats.
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* For any specified "TIAS" bandwidth value, set this value as a
constraint on the maximum RTP bitrate to be used when sending
media, as specified in [RFC3890]. If a "TIAS" value is not
present, but an "AS" value is specified, generate a "TIAS"
value using this formula:
TIAS = AS * 1000 * 0.95 - 50 * 40 * 8
The 50 is based on 50 packets per second, the 40 is based on an
estimate of total header size, the 1000 changes the unit from
kbps to bps (as required by TIAS), and the 0.95 is to allocate
5% to RTCP. If more accurate control of bandwidth is needed,
"TIAS" should be used instead of "AS".
* For any "RR" or "RS" bandwidth values, handle as specified in
[RFC3556], Section 2.
* Any specified "CT" bandwidth value MUST be ignored, as the
meaning of this construct at the media level is not well
defined.
* If the media section is of type audio:
+ For each specified "CN" media format, enable DTX for all
supported media formats with the same clockrate, as
described in [RFC3389], Section 5, except for formats that
have their own internal DTX mechanisms. DTX for such
formats (e.g., Opus) is controlled via fmtp parameters, as
discussed in Section 5.2.3.2.
+ For each specified "telephone-event" media format, enable
DTMF transmission for all supported media formats with the
same clockrate, as described in [RFC4733], Section 2.5.1.2.
If the application attempts to transmit DTMF when using a
media format that does not have a corresponding telephone-
event format, this MUST result in an error.
+ For any specified "ptime" value, configure the available
media formats to use the specified packet size. If the
specified size is not supported for a media format, use the
next closest value instead.
Finally, if this description is of type "pranswer" or "answer",
follow the processing defined in the Section 5.10 section below.
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In addition to the steps mentioned above for processing a local or
remote description, the following steps are performed when processing
a description of type "pranswer" or "answer".
For each media section, the following steps MUST be performed:
o If the media section has been rejected (i.e. port is set to zero
in the answer), stop any reception or transmission of media for
this section, and discard any associated ICE components, as
described in Section 9.2.1.3 of [RFC5245].
o If the remote DTLS fingerprint has been changed or the dtls-id has
changed, tear down the DTLS connection. If a DTLS connection
needs to be torn down but the answer does not indicate an ICE
restart, an error MUST be generated. If an ICE restart is
performed without a change in dtls-id or fingerprint, then the
same DTLS connection is continued over the new ICE channel.
o If no valid DTLS connection exists, prepare to start a DTLS
connection, using the specified roles and fingerprints, on any
underlying ICE components, once they are active.
o If the media section proto value indicates use of RTP:
* If the media section references any media formats, RTP header
extensions, or RTCP feedback mechanisms that were not present
in the corresponding media section in the offer, this indicates
a negotiation problem and MUST result in an error.
* If the media section has RTCP mux enabled, discard any RTCP
component, and begin or continue muxing RTCP over the RTP
component, as specified in [RFC5761], Section 5.1.3.
Otherwise, prepare to transmit RTCP over the RTCP component; if
no RTCP component exists, because RTCP mux was previously
enabled, this MUST result in an error.
* If the media section has reduced-size RTCP enabled, configure
the RTCP transmission for this media section to use reduced-
size RTCP, as specified in [RFC5506].
* If the directional attribute in the answer is of type
"sendrecv" or "sendonly", choose the media format to send as
the most preferred media format from the remote description
that is also present in the answer, as described in [RFC3264],
Sections 6.1 and 7, and start transmitting RTP media once the
underlying transport layers have been established. If a SSRC
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has not already been chosen for this outgoing RTP stream,
choose a random one.
* The payload type mapping from the remote description is used to
determine payload types for the outgoing RTP streams, including
the payload type for the send media format chosen above. Any
RTP header extensions that were negotiated should be included
in the outgoing RTP streams, using the extension mapping from
the remote description; if the RID header extension has been
negotiated, and RID values are specified, include the RID
header extension in the outgoing RTP streams, as indicated in
[I-D.ietf-mmusic-rid], Section 4.
* If simulcast has been negotiated, send the number of Source RTP
Streams as specified in [I-D.ietf-mmusic-sdp-simulcast],
Section 6.2.2.
* If the send media format chosen above has a corresponding "rtx"
media format, or a FEC mechanism has been negotiated, establish
a Redundancy RTP Stream with a random SSRC for each Source RTP
Stream, and start or continue transmitting RTX/FEC packets as
needed.
* If the send media format chosen above has a corresponding "red"
media format of the same clockrate, allow redundant encoding
using the specified format for resiliency purposes, as
discussed in [I-D.ietf-rtcweb-fec], Section 3.2. Note that
unlike RTX or FEC media formats, the "red" format is
transmitted on the Source RTP Stream, not the Redundancy RTP
Stream.
* Enable the RTCP feedback mechanisms referenced in the media
section for all Source RTP Streams using the specified media
formats. Specifically, begin or continue sending the requested
feedback types and reacting to received feedback, as specified
in [RFC4585], Section 4.2. When sending RTCP feedback, use the
SSRC of an outgoing Source RTP Stream as the RTCP sender SSRC;
if no outgoing Source RTP Stream exists, choose a random one.
* If the directional attribute is of type "recvonly" or
"inactive", stop transmitting all RTP media, but continue
sending RTCP, as described in [RFC3264], Section 5.1.
o If the media section proto value indicates use of SCTP:
* If no SCTP association yet exists, prepare to initiate a SCTP
association over the associated ICE component and DTLS
connection, using the local SCTP port value from the local
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description, and the remote SCTP port value from the remote
description, as described in [I-D.ietf-mmusic-sctp-sdp],
Section 10.2.
If the answer contains valid bundle groups, discard any ICE
components for the m= sections that will be bundled onto the primary
ICE components in each bundle, and begin muxing these m= sections
accordingly, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.2.
6. Processing RTP/RTCP packets
Note: The following algorithm does not yet have WG consensus but is
included here as something concrete for the working group to discuss.
When an RTP packet is received by a transport and passes SRTP
authentication, that packet needs to be routed to the correct
RtpReceiver. For each transport, the following steps MUST be
followed to prepare to route packets:
Construct a table mapping MID to RtpReceiver for each RtpReceiver
configured to receive from this transport.
Construct a table mapping incoming SSRC to RtpReceiver for each
RtpReceiver configured to receive from this transport and for each
SSRC that RtpReceiver is configured to receive. Some of the SSRCs
may be present in the m= section corresponding to that RtpReceiver
in the remote description.
Construct a table mapping outgoing SSRC to RtpSender for each
RtpSender configured to transmit from this transport and for each
SSRC that RtpSender is configured to use when sending.
Construct a table mapping payload type to RtpReceiver for each
RtpReceiver configured to receive from this transport and for each
payload type that RtpReceiver is configured to receive. The
payload types of a given RtpReceiver are found in the m= section
corresponding to that RtpReceiver in the local description. If
any payload type could map to more than one RtpReceiver, map to
the RtpReceiver whose m= section appears earliest in the local
description.
As RtpTransceivers (and, thus, RtpReceivers) are added, removed,
stopped, or reconfigured, the tables above must also be updated.
For each RTP packet received, the following steps MUST be followed to
route the packet:
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If the packet has a MID and that MID is not in the table mapping
MID to RtpReceiver, drop the packet and stop.
If the packet has a MID and that MID is in the table mapping MID
to RtpReceiver, update the incoming SSRC mapping table to include
an entry that maps the packet's SSRC to the RtpReceiver for that
MID.
If the packet's SSRC is in the incoming SSRC mapping table,
deliver the packet to the associated RtpReceiver and stop.
If the packet's payload type is in the payload type table, update
the the incoming SSRC mapping table to include an entry that maps
the packet's SSRC to the RtpReceiver for that payload type. In
addition, deliver the packet to the associated RtpReceiver and
stop.
Otherwise, drop the packet.
For each RTCP packet received (including each RTCP packet that is
part of a compound RTCP packet), the following type-specific handling
MUST be performed to route the packet:
If the packet is of type SR, and the sender SSRC for the packet is
found in the incoming SSRC table, deliver a copy of the packet to
the RtpReceiver associated with that SSRC. In addition, for each
report block in the report whose SSRC is found in the outgoing
SSRC table, deliver a copy of the RTCP packet to the RtpSender
associated with that SSRC.
If the packet is of type RR, for each report block in the packet
whose SSRC is found in the outgoing SSRC table, deliver a copy of
the RTCP packet to the RtpSender associated with that SSRC.
If the packet is of type SDES, and the sender SSRC for the packet
is found in the incoming SSRC table, deliver the packet to the
RtpReceiver associated with that SSRC. In addition, for each
chunk in the packet that contains a MID that is in the table
mapping MID to RtpReceiver, update the incoming SSRC mapping table
to include an entry that maps the SSRC for that chunk to the
RtpReceiver associated with that MID. (This case can occur when
RTCP for a source is received before any RTP packets.)
If the packet is of type BYE, for each SSRC indicated in the
packet that is found in the incoming SSRC table, deliver a copy of
the packet to the RtpReceiver associated with that SSRC.
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If the packet is of type RTPFB or PSFB, as defined in [RFC4585],
and the media source SSRC for the packet is found in the outgoing
SSRC table, deliver the packet to the RtpSender associated with
that SSRC.
After packets are routed to the RtpReceiver, further processing of
the RTP packets is done at the RtpReceiver level. This includes
using [I-D.ietf-mmusic-rid] to distinguish between multiple Encoded
Streams, as well as determine which Source RTP stream should be
repaired by a given Redundancy RTP stream. If the RTP packet's PT
does not match any codec in use by the RtpReceiver, the packet will
be dropped.
7. Examples
Note that this example section shows several SDP fragments. To
format in 72 columns, some of the lines in SDP have been split into
multiple lines, where leading whitespace indicates that a line is a
continuation of the previous line. In addition, some blank lines
have been added to improve readability but are not valid in SDP.
More examples of SDP for WebRTC call flows can be found in
[I-D.nandakumar-rtcweb-sdp].
7.1. Simple Example
This section shows a very simple example that sets up a minimal audio
/ video call between two browsers and does not use trickle ICE. The
example in the following section provides a more realistic example of
what would happen in a normal browser to browser connection.
The flow shows Alice's browser initiating the session to Bob's
browser. The messages from Alice's JS to Bob's JS are assumed to
flow over some signaling protocol via a web server. The JS on both
Alice's side and Bob's side waits for all candidates before sending
the offer or answer, so the offers and answers are complete. Trickle
ICE is not used. Both Alice and Bob are using the default policy of
balanced.
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c=IN IP4 192.0.2.2
a=rtcp 20001 IN IP4 192.0.2.2
a=mid:v1
a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1v0
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active
a=rtcp-mux
a=rtcp-rsize
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
7.2. Normal Examples
This section shows a typical example of a session between two
browsers setting up an audio channel and a data channel. Trickle ICE
is used in full trickle mode with a bundle policy of max-bundle, an
RTCP mux policy of require, and a single TURN server. Later, two
video flows, one for the presenter and one for screen sharing, are
added to the session. This example shows Alice's browser initiating
the session to Bob's browser. The messages from Alice's JS to Bob's
JS are assumed to flow over some signaling protocol via a web server.
// set up local media state
AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with an audio track
AliceJS->AliceUA: createDataChannel to get data channel
AliceJS->AliceUA: createOffer to get |offer-B1|
AliceJS->AliceUA: setLocalDescription with |offer-B1|
// |offer-B1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-B1|
WebServer->BobJS: signaling with |offer-B1|
// |offer-B1| arrives at Bob
BobJS->BobUA: create a PeerConnection
BobJS->BobUA: setRemoteDescription with |offer-B1|
BobUA->BobJS: onaddstream with audio track from Alice
// candidates are sent to Bob
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a=setup:passive
a=rtcp-mux
a=rtcp-rsize
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
8. Security Considerations
The IETF has published separate documents
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security] describing
the security architecture for WebRTC as a whole. The remainder of
this section describes security considerations for this document.
While formally the JSEP interface is an API, it is better to think of
it is an Internet protocol, with the JS being untrustworthy from the
perspective of the browser. Thus, the threat model of [RFC3552]
applies. In particular, JS can call the API in any order and with
any inputs, including malicious ones. This is particularly relevant
when we consider the SDP which is passed to setLocalDescription().
While correct API usage requires that the application pass in SDP
which was derived from createOffer() or createAnswer(), there is no
guarantee that applications do so. The browser MUST be prepared for
the JS to pass in bogus data instead.
Conversely, the application programmer MUST recognize that the JS
does not have complete control of browser behavior. One case that
bears particular mention is that editing ICE candidates out of the
SDP or suppressing trickled candidates does not have the expected
behavior: implementations will still perform checks from those
candidates even if they are not sent to the other side. Thus, for
instance, it is not possible to prevent the remote peer from learning
your public IP address by removing server reflexive candidates.
Applications which wish to conceal their public IP address should
instead configure the ICE agent to use only relay candidates.
9. IANA Considerations
This document requires no actions from IANA.
10. Acknowledgements
Significant text incorporated in the draft as well and review was
provided by Peter Thatcher, Taylor Brandstetter, Harald Alvestrand
and Suhas Nandakumar. Dan Burnett, Neil Stratford, Anant Narayanan,
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o Specify how SDP session version in o= line.
o Require that when doing an re-offer, the capabilities of the new
session are mostly required to be a subset of the previously
negotiated session.
o Clarified ICE restart interaction with bundle-only.
o Remove support for changing SDP before calling
setLocalDescription.
o Specify algorithm for demuxing RTP based on MID, PT, and SSRC.
o Clarify rules for rejecting m= lines when bundle policy is
balanced or max-bundle.
Changes in draft-15:
o Clarify text around codecs offered in subsequent transactions to
refer to what's been negotiated.
o Rewrite LS handling text to indicate edge cases and that we're
living with them.
o Require that answerer reject m= lines when there are no codecs in
common.
o Enforce max-bundle on offer processing.
o Fix TIAS formula to handle bits vs. kilobits.
o Describe addTrack algorithm.
o Clean up references.
Changes in draft-14:
o Added discussion of RtpTransceivers + RtpSenders + RtpReceivers,
and how they interact with createOffer/createAnswer.
o Removed obsolete OfferToReceiveX options.
o Explained how addIceCandidate can be used for end-of-candidates.
Changes in draft-13:
o Clarified which SDP lines can be ignored.
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